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1. SNMP Trap Destinations Page essed SNMP Community Strings Page SNMP V3 pati Page Regional a Page EE GEOL GHEE RAE RARE SAAS Maintenance Actions Page Reset Confirmation Message Box Software Upgrade Key with Mul 5 usm zeti AO rt Software Upgrade Wizard SOTG6 cc cscsesscancesvesnansnverinacteveranssouieneeaormracasieroncius IOS Figure 3 106 End Process Wizard Page seiccccdencieoasacnecisnnprncacsennnsamavbnennsenntunaemunioenernanane dT Fiotir 3 109 Configuration File PAGE u oa Bd b lo n K PA kb KS oka ov n LEM Figure 3 110 Message Log Screen Figure 3 111 Figure 3 112 Figure 3 113 Active Alarms Page Figure 3 114 1 Ethernet Port Information P cede asp O eat ied ae R 174 Penomanbe Statistics POE iso na dea rob Kao aaa ad boud k LEE Calls Count Page Sasi ascent icc dea a son tat SIP User s Manual 10 Document LTRT 65413 SIP User s Manual Contents n and Reset Version 6 0 11 March 2010 7a i 4 wl AudioCodes MediaPack Series List of Tables Table 1 1 Supported MediaPack Series Configurations c ccccccscccsscssscssecesscessecsseccssscsseessessssesee TL Description of Toolbar Buttons 2 ini File Parameter for Welcome Login Mess j 3 Description of the Areas of the Home Page 3 4 Multiple Interface Table Parameters Description gt NFS Settings Parameters
2. 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 ff March 2010 ca AudioCodes 3 3 3 6 Configuring the General Security Settings MediaPack Series The General Security Settings page is used to configure various security features For a description of the parameters appearing on this page refer Configuration Parameters Reference on page 207 gt To configure the general security parameters 1 Open the General Security Settings page Configuration tab gt Security Settings menu gt General Security Settings page item Figure 3 53 General Security Settings Page P HTTP Authentication Mode Secured Web Connection HTTPS Voice Menu Password Digest When Possible HTTP and HTTPS 12345 wv General RADIUS Setting Enable RADIUS Access Control Use RADIUS for Web Telnet Login RADIUS Authentication Server IP Address RADIUS Authentication Server Port RADIUS Shared Secret Disable Disable 0 0 0 0 1645 wv General RADIUS Authentication Default Access Level Device Behavior Upon RADIUS Timeout Local RADIUS Password Cache Mode RADIUS VSA Vendor ID RADIUS VSA Access Level Attribute Local RADIUS Password Cache Timeout sec 200 Verify Access Locally Reset Ti
3. Internal Call Digit Pattern External Call Digit Pattern Disconnect Call Digit Pattern Digit To Ignore Digit Pattern Message Waiting Indication MWI MWI Off Digit Pattern MWI On Digit Pattern o MWI Suffix Pattern MWI Source Number v SMDI Enable SMDI Disable v SMDI Timeout msec 2000 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 149 March 2010 7a L tal AudioCodes MediaPack Series 3 3 5 2 Configuring RADIUS Accounting Parameters The RADIUS Parameters page is used for configuring the Remote Authentication Dial In User Service RADIUS accounting parameters For a description of these parameters refer to Configuration Parameters Reference on page 207 gt To configure the RADIUS parameters 1 Open the RADIUS Parameters page Configuration tab gt Advanced Applications menu gt RADIUS Parameters page item Figure 3 93 RADIUS Parameters Page v Enable RADIUS Access Control Disable Accounting Server IP Address 0 0 0 0 Accounting Port 1646 RADIUS Accounting Type At Call Release 444 Indications None 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To s
4. SNMP Trusted Manager 5 0 0 0 0 3 Select the check box corresponding to the SNMP Trusted Manager that you want to enable and for whom you want to define an IP address 4 Define an IP address in dotted decimal notation 5 Click the Submit button to apply your changes 6 To save the changes refer to Saving Configuration on page 161 3 4 1 2 Configuring the Regional Settings The Regional Settings page allows you to define and view the device s internal date and time gt To configure the device s date and time 1 Open the Regional Settings page Management tab gt Management Configuration menu gt Regional Settings page item Figure 3 100 Regional Settings Page Minutes Seconds 16 23 2 Enter the current date and time in the geographical location in which the device is installed 3 Click the Submit button the date and time are automatically updated If the device is configured to obtain the date and time from an SNTP server refer to Configuring the Application Settings on page 54 the fields on this page are read only and cannot be modified For an explanation on SNTP refer to Simple Network Time Protocol Support on page 447 After performing a hardware reset the date and time are returned to their defaults and therefore should be updated SIP User s Manual 158 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 4 1 3 Maintenance Action
5. Table 3 36 IP Connectivity Parameters Column Name Description IP Address The IP address can be one of the following P address defined as the destination IP address in the Tel to IP Routing IP address resolved from the host name defined as the destination IP address in the Tel to IP Routing Host Name Host name or IP address as defined in the Tel to IP Routing Connectivity The method according to which the destination IP address is gueried Method periodically ICMP ping or SIP OPTIONS reguest Version 6 0 183 March 2010 A e AudioCodes MediaPack Series Column Name Connectivity Status Quality Status Quality Info DNS Status SIP User s Manual Description The status of the IP address connectivity according to the method in the Connectivity Method field OK Remote side responds to periodic connectivity queries Lost Remote side didn t respond for a short period Fail Remote side doesn t respond Init Connectivity queries not started e g IP address not resolved Disable The connectivity option is disabled i e parameter Alt Routing Tel to IP Mode AltRoutingTel2IPMode ini is set to None or QoS Determines the QoS according to packet loss and delay of the IP address Unknown Recent quality information isn t available OK Poor Notes This parameter is applicable only if the parameter Alt Routing Tel to IP Mode is set to QoS or
6. red line not connected only applicable to FXO devices grey channel inactive blue handset is off hook green active RTP stream You can also view the channel s port settings refer to Viewing Port Information on page 49 reset the port refer to Resetting an Analog Channel on page 48 and assign a name to the port refer to Assigning a Port Name on page 48 If clicked the Ethernet Port Information page opens displaying Ethernet port configuration settings refer to Viewing Ethernet Port Information on page 173 Currently not supported Currently not supported Always lit green indicating power received by the device Assigning a Port Name The Home page allows you to assign an arbitrary name or a brief description to each port This description appears as a tooltip when you move your mouse over the port To add a port description Click the reguired port icon a shortcut menu appears as shown below Figure 3 28 Shortcut Menu Example MP 11x Uplink From the shortcut menu choose Update Port Info a text box appears Figure 3 29 Text Box for Port Name Example MP 11x T Portneme ordescrpfon ApplyPortlnfo 3 2 2 SIP User s Manual 3 Type a brief description for the port and then click Apply Port Info Resetting an Analog Channel The Home page allows you to inactivate reset an FXO or FXS analog channel
7. Only delete your PC s IP address last from the Web amp Telnet Access List page If it s deleted before the last access from your PC is denied after it s deleted 3 3 3 3 Configuring the Firewall Settings The device provides an internal firewall allowing you the security administrator to define network traffic filtering rules You can add up to 50 ordered firewall rules The access list provides the following features Block traffic from known malicious sources Only allow traffic from known friendly sources and block all others Mix allowed and blocked network sources Limit traffic to a pre defined rate blocking the excess Limit traffic to specific protocols and specific port ranges on the device For each packet received on the network interface the table is scanned from the top down until a matching rule is found This rule can either deny block or permit allow the packet Once a rule in the table is located subsequent rules further down the table are ignored If the end of the table is reached without a match the packet is accepted For detailed information on the internal firewall refer to the Product Reference Manual Note You can also configure the firewall settings using the ini file table parameter Access List refer to Security Parameters on page 232 gt To add firewall rules 1 Open the Firewall Settings page Configuration tab gt Security Settings menu gt Firewall Settings pag
8. K tal AudioCodes MediaPack Series The received INVITE message is routed as depicted in the flow chart below Figure 9 2 SAS Routing in Emergency Mode Receiving INVITE request z Check the SAS DB for internal registered users SE Y Check for online redundant SAS Send INVITE to server eNO Foundiusers ES registered user Via limitations Y Try to route according to 4 NO Found server YES Routing Table Send INVITE to redundant SAS Y Send INVITE YES according to Routing Table Send INVITE to 4 NO Found entry default GW in table 9 2 1 1 Configuring SAS For configuring the device to operate with SAS perform the following configurations IsProxyUsed 1 ProxylP 0 lt SAS agent s IP address i e the device gt ProxylP 1 lt external Proxy server IP address gt IsRegisterNeeded 1 for the device RegistrarlP SIPDestinationPort 5080 IsUserPhone 0 don t use user phone in SIP URL IsUserPhonelnFrom 0 don t use user phone in From Header IsFallbackUsed 0 EnableProxyKeepAlive 1 enables keep alive with Proxy using OPTIONS EnableSAS 1 SASLocalSIPUDPPort default 5080 SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 E SASDefaultGatewaylP lt SAS gateway IP address gt SIP User s Manual 382 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities m
9. Note This feature is applicable only to FXS interfaces Version 6 0 305 March 2010 Ao tal AudioCodes MediaPack Series 6 8 10 Automatic Dialing Parameters The automatic dialing upon off hook parameters are described in the table below Parameter Table 6 43 Automatic Dialing Parameters Description Web Automatic Dialing Table EMS SIP Endpoints gt Auto Dial TargetOfChannel SIP User s Manual This ini file table parameter defines telephone numbers that are automatically dialed when a specific FXS or FXO port is used i e telephone is off hooked The format of this parameter is as follows TargetOfChannel FORMAT TargetOfChannel_Index TargetOfChannel Destination TargetOfChannel_Type TargetOfChannel Where Index Port number where 0 depicts Port 1 Destination Destination phone number that you want dialed Type v 0 Disable automatic dialing is disabled v 1 Enable Destination phone number is automatically dialed if phone is off hooked for FXS interface or ring signal is applied to port FXO interface v 2 Hotline enables the Hotline feature where if the phone is off hooked and no digit is pressed for a user defined duration configured by the parameter HotLine ToneDuration the destination phone number is automatically dialed For example the below configuration defines automatic dialing of phone number 911 when the phone that is connected to Port 1 is off
10. Tonelndex Where FXSPort_First starting range of FXS ports FXSPort_Last end range of FXS ports SourcePrefix prefix of the calling number Prioritylndex index for Distinctive Ringing and Call Waiting tones default is 0 v Ringing tone index index in the CPT file for playing the ring tone v Call Waiting tone index priority index FirstCallWaitingTonelD For example if you want to select the Call Waiting tone defined in the CPT file at Index 9 then you can enter 1 as the priority index and the value 8 for FirstCallWaitingTonelD The summation of these values equals 9 i e index 9 For example the configuration below plays the tone Index 3 to FXS ports 1 and 2 if the source number prefix of the received call is 20 Tonelndex 1 1 2 20 3 Notes You can define up to 50 indices This parameter is applicable only to FXS interfaces Typically the Ringing and or Call Waiting tone played is indicated in the SIP Alert Info header field of the received INVITE message If this header is not present then the tone played is according to the settings in this table For depicting a range of FXS ports use the syntax x y e g 1 4 for ports 1 through 4 SIP User s Manual 318 Document LTRT 65413 SIP User s Manual Parameter Web EMS Dial Tone Duration sec TimeForDialTone Web EMS Stutter Tone Duration StutterToneDuration Web FXO AutoDial Play BusyTone EM
11. Determines the routing mode after a call redirection i e a 3xx SIP response is received or transfer i e a SIP REFER reguest is received 0 Standard INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response default 1 Proxy Sends a new INVITE to the Proxy Note This option is applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0 266 Document LTRT 65413 SIP User s Manual Parameter Web EMS DNS Query Type DNSQueryType Version 6 0 6 Configuration Parameters Reference Description 2 Routing Table Uses the Routing table to locate the destination and then sends a new INVITE to this destination Notes When this parameter is set to 1 and the INVITE sent to the Proxy fails the device re routes the call according to the Standard mode 0 When this parameter is set to 2 and the INVITE fails the device re routes the call according to the Standard mode 0 If DNS resolution fails the device attempts to route the call to the Proxy If routing to the Proxy also fails the Redirect Transfer request is rejected When this parameter is set to 2 the XferPrefix parameter can be used to define different routing rules for redirect calls This parameter is disregarded if the parameter AlwaysSendToProxy is s
12. Determines the use of Tel Source Number and Display Name for Tel to IP calls 0 No Ifa Tel Display Name is received the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name If no Display Name is received from the Tel side the IP Display Name remains empty default 1 Yes If a Tel Display Name is received the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name If no Display Name is received from the Tel side the Tel Source Number is used as the IP Source Number and also as the IP Display Name 2 Overwrite The Tel Source Number is used as the IP Source Number and also as the IP Display Name even if the received Tel Display Name is not empty 333 March 2010 ca AudioCodes Parameter Web EMS Use Display Name as Source Number UseDisplayNameAsSourceNum ber Web Use Routing Table for Host Names and Profiles EMS Use Routing Table For Host Names AlwaysUseRouteTable Web EMS Tel to IP Routing Mode RouteModeTel2IP Web Tel to IP Routing EMS SIP Routing gt Tel to IP Prefix SIP User s Manual MediaPack Series Description Determines the use of Source Number and Display Name for IP to Tel calls 0 No If IP Display Name is received the IP Source Number is used as the Tel Source Number and the IP Display Name is used as the Tel Display Name If no Display Name is re
13. EMS ETSI VMWI Type One Standard ETSIVMWITypeOneStandard EMS Bellcore VMWI Type One Standard BellcoreVMWITypeOneStandard SIP User s Manual MediaPack Series Description Determines the transport layer used for outgoing SIP dialogs initiated by the device to the MWI server 1 Not Configured default 0 UDP 1 TCP 2 TLS Note When set to Not Configured the value of the parameter SIPTransportType is used The MWI subscription expiration time in seconds The default is 7200 seconds The range is 10 to 2 000 000 Subscription retry time in seconds after last subscription failure The default is 120 seconds The range is 10 to 2 000 000 Determines the method the device uses to subscribe to an MWI server 0 Per Endpoint Each endpoint subscribes separately typically used for FXS interfaces default 1 Per Gateway Single subscription for the entire device typically used for FXO interfaces Selects the ETSI Visual Message Waiting Indication VMWI Type 1 sub standard 0 ETSI VMWI between rings default 1 ETSI VMWI before ring DT AS 2 ETSI VMWI before ring RP AS 3 ETSI VMWI before ring LR DT AS 4 ETSI VMWI not ring related DT AS 5 ETSI VMWI not ring related RP AS 6 ETSI VMWI not ring related LR DT AS Note For this parameter to take effect a device reset is reguired Selects the Bellcore VMWI sub standard 0 Between
14. FXO interfaces If a ring signal is detected the device seizes the FXO line plays a dial tone and then waits for DTMF digits If no digits are detected for a user defined time configured using the parameter HotLineToneDuration the number in the Destination Phone Number field is automatically dialed by sending a SIP INVITE message with this number 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 9 3 Configuring Caller Display Information The Caller Display Information page allows you to enable the device to send Caller ID information to IP when a call is made The called party can use this information for caller identification The information configured in this page is sent in an INVITE message in the From header For information on Caller ID restriction according to destination source prefixes refer to Configuring the Number Manipulation Tables on page 115 SIP User s Manual 138 Document LTRT 65413 SIP User s Manual 3 Web Based Management Version 6 0 To configure the Caller Display Information Open the Caller Display Information page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Caller Display Information page item Figure 3 86 Caller Display Information Page canes Caller ID Name Presentation FXS Private Restricted w FXS Susan C Restrict
15. If the device was originally operating in HTTPS mode and you disabled it in Step 2 then return it to HTTPS by setting the parameter Secured Web Connection HTTPS to HTTPS Only 1 refer to Configuring the General Security Settings on page 78 The certificate replacement process can be repeated when necessary e g the new certificate expires It is possible to use the IP address of the device e g 10 3 3 1 instead of a qualified DNS name in the Subject Name This is not recommended since the IP address is subject to changes and may not uniquely identify the device The server certificate can also be loaded via ini file using the parameter HTTPSCertFileName SIP User s Manual 74 Document LTRT 65413 SIP User s Manual 3 Web Based Management To apply the loaded certificate for IPSec negotiations Open the IKE Table page refer to Configuring the IP Security Proposal Table on page 79 the Loaded Certificates Files group lists the newly uploaded certificates as shown below Figure 3 51 IKE Table Listing Loaded Certificate Files Loaded Certificate Files Fourth Proposal DH Group Not Detned 2 First Proposal Encryption Type Trple DES CBC First Proposal Authentication Type HMAC SHA4 1 S6 First Proposal DH Group DH 1024 B1T Second Proposal Encryption Type Not Defined Second Proposal Authentication Type Not Defined Second Proposel OH Group Not Defined Third Proposal Enecrypton Type Not Defined T
16. Note To use this service the devices at both ends must support this option Web Call Forwarding Table EMS SIP Endpoints gt Call Forward Fwdlnfo This ini file table parameter forwards redirects IP to Tel calls using SIP 302 response to other device ports or an IP destination based on the device s port to which the call was originally routed The format of this parameter is as follows Fwdinfo FORMAT Fwdlnfo_Index Fwdlinfo_Type FwdInfo Destination FwdlInfo_NoReplyTime Fwdlnfo Where Index Port number where 0 depicts Port 1 Type the scenario for forwarding the call 0 Deactivate Don t forward incoming calls default 1 On Busy Forward incoming calls when the port is busy 2 Unconditional Always forward incoming calls 3 No Answer Forward incoming calls that are not answered within the time specified in the Time for No Reply Forward field 4 On Busy or No Answer Forward incoming calls when the port is busy or when calls are not answered within the time specified in the Time for No Reply Forward field v 5 Do Not Disturb Immediately reject incoming calls Destination Telephone number or URI lt number gt lt IP address gt to where the call is forwarded se a Ska NoReplyTime Timeout in seconds for No Reply If you have set the Forward Type for this port to No Answer 3 enter the number of seconds the device waits before forwarding the call to th
17. This section provides a brief description on configuring various device configurations using AudioCodes Element Management System EMS The EMS is an advanced solution for standards based management of gateways within VoP networks covering all areas vital for the efficient operation administration management and provisioning OAM amp P of AudioCodes families of gateways The EMS enables Network Equipment Providers NEPs and System Integrators Sls the ability to offer customers rapid time to market and inclusive cost effective management of next generation networks The standards compliant EMS uses distributed SNMP based management software optimized to support day to day Network Operation Center NOC activities offering a feature rich management framework It supports fault management configuration and security For a detailed description of the EMS tool refer to the EMS User s Manual and EMS Server IOM Manual Familiarizing yourself with EMS GUI The areas of the EMS graphical user interface GUI are shown in the figure below Figure 5 1 Areas of the EMS GUI Fae View Tools Faults Security Help MO Tree MG Node Info Navigation Configuration Alarms Performance O M Z O indeter 17 2106 Feb 2320101 SP Ostewsy EMS Server Otndeter 17 2045 Feb 23 20001 SP EMS Server Olndeter 171231 Feb 2320101 SP OMS Server The MG Tree is a hierarchical tree like structure that lists all the devices managed by EMS The tree in
18. e RxDTMFOption 3 e TxDTMFOption 4 Note that to set the RFC 2833 payload type with a different value other than its default configure the RFC2833PayloadType parameter The device negotiates the RFC 2833 payload type using local and remote SDP and sends packets using the payload type from the received SDP The device expects to receive RFC 2833 packets with the same payload type as configured by the RFC2833PayloadType parameter If the remote side doesn t include telephony event in its SDP the device sends DTMF digits in transparent mode as part of the voice stream Sending DTMF digits in RTP packets as part of the audio stream DTMF Relay is disabled This method is typically used with G 711 coders with other low bit rate LBR coders the quality of the DTMF digits is reduced To enable this mode define the following e RxDTMFOption 0 i e disabled e TxDTMFOption 0 i e disabled e DTMFTransportType 2 i e transparent SIP User s Manual 384 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities m Using INFO message according to Korea mode DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 i e disabled e TxDTMFOption 3 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 The device is always ready to receive DTMF packets over IP in all possible
19. Description Defines the response of the device upon receipt of a SIP 183 response 0 Progress A 183 response without SDP does not cause the device to play a ringback tone default 1 Alert 183 response is handled by the device as if a 180 Ringing response is received and the device plays a ringback tone Determines the numerical value that is sent in the Session Expires header in the first INVITE reguest or response if the call is answered The valid range is 1 to 86 400 sec The default is 0 i e the Session Expires header is disabled Defines the time in seconds that is used in the Min SE header This header defines the minimum time that the user agent refreshes the session The valid range is 10 to 100 000 The default value is 90 Determines the SIP method used for session timer updates 0 Re INVITE Uses Re INVITE messages for session timer updates default 1 UPDATE Uses UPDATE messages Notes The device can receive session timer refreshes using both methods The UPDATE message used for session timer is excluded from the SDP body Determines whether the device removes the to header tag from final SIP failure responses to INVITE transactions 0 Do not remove tag default 1 Remove tag Enables or disables the use of the rtcp attribute in the outgoing SDP 0 Disable default 1 Enable Defines the user part value of the Request UR
20. Determines the Q 850 cause value specified in the SIP Reason header that is included in a 4xx response when a Special Information Tone SIT is detected on an IP to Tel call The valid range is 0 to 127 The default value is 34 Notes For mapping specific SIT tones you can use the SITQ850CauseForNC SITQ850CauseForlC SITQ850CauseForVC and SITQ850CauseForRO parameters This parameter is applicable only to FXO interfaces Determines the Q 850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT NC No Circuit Found Special Information Tone is detected from the Tel for IP to Tel calls 260 Document LTRT 65413 SIP User s Manual Parameter Web EMS SIT Q850 Cause For IC SITQ850CauseForlC Web EMS SIT Q850 Cause For VC SITQ850CauseForVC Web EMS SIT Q850 Cause For RO SITQ850CauseForRO 6 Configuration Parameters Reference Description The valid range is 0 to 127 The default value is 34 Notes When not configured i e default the SITQ850Cause parameter is used This parameter is applicable only to FXO interfaces Determines the Q 850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT IC Operator Intercept Special Information Tone is detected from the Tel for IP to Tel calls The valid range is 0 to 127 The default value is 1 not configured Notes When not configured i e default the SITQ8
21. Determines the authentication mode for the Web interface 0 Basic Mode Basic authentication clear text is used default 1 Digest When Possible Digest authentication MD5 is used 2 Basic if HTTPS Digest if HTTP Digest authentication MD5 is used for HTTP and basic authentication is used for HTTPS Note When RADIUS login is enabled i e the parameter WebRADIUSLogin is set to 1 basic authentication is forced 233 March 2010 ca AudioCodes Parameter HTTPSReguireClientCertificate HTTPSRootFileName HTTPSPkeyFileName HTTPSCertFileName 6 4 3 SRTP Parameters MediaPack Series Description Reguires client certificates for HTTPS connection The client certificate must be preloaded to the device and its matching private key must be installed on the managing PC Time and date must be correctly set on the device for the client certificate to be verified 0 Client certificates are not required default 1 Client certificates are required Note For this parameter to take effect a device reset is required Defines the name of the HTTPS trusted root certificate file to be loaded using TFTP The file must be in base64 encoded PEM Privacy Enhanced Mail format The valid range is a 47 character string Note This parameter is only applicable when the device is loaded using BootP TFTP For information on loading this file using the Web interface refer to the Product R
22. For example Authentication 0 john 1325 user name john with password 1325 for authenticating Port 1 Authentication 1 lee 1552 user name lee with password 1552 for authenticating Port 2 Notes The parameter AuthenticationMode determines whether authentication is performed per port or for the entire device If authentication is performed for the entire device the configuration in this table parameter is ignored If the user name or password are not configured the port s phone number configured using the parameter TrunkGroup Endpoint Phone Number table and global password using the individual parameter Password are used for authentication Authentication is typically used for FXS interfaces but can also be used for FXO interfaces For configuring the Authentication table using the Web interface refer to Configuring Authentication on page 136 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 SIP User s Manual 264 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web Account Table EMS SIP Endpoints gt Account Account Proxy Registration Parameters Web Use Default Proxy EMS Proxy Used IsProxyUsed Web EMS Proxy Name ProxyName Version 6 0 Description This ini file table parameter configures the Account table for registering and or authenticating dige
23. Gateway Name TrunkGroupSettings GatewayNa me Contact User TrunkGroupSettings ContactUse r Version 6 0 3 Web Based Management Description Notes To enable Hunt Group registrations configure the global parameter IsRegisterNeeded to 1 This is unnecessary for Per Account registration mode f no mode is selected the registration is performed according to the global registration parameter ChannelSelectMode If the device is configured globally ChannelSelectMode to register Per Endpoint and a endpoints Group comprising four FXO endpoints is configured to register Per Gateway the device registers all endpoints except the first four endpoints The endpoints Group of these four endpoints sends a single registration request The Serving IP Group ID to where INVITE messages initiated by this Hunt Group s endpoints are sent The actual destination to where these INVITE messages are sent is according to the Proxy Set ID refer to Configuring the Proxy Sets Table on page 97 associated with this Serving IP Group The Request URI hostname in the INVITE and REGISTER messages except for Per Account registration modes is set to the value of the field SIP Group Name defined in the IP Group table refer to Configuring the IP Groups on page 91 If no Serving IP Group ID is selected the INVITE messages are sent to the default Proxy or according to the Tel to IP Routing refer to Configuring the T
24. LTRT 65413 SIP User s Manual 5 Element Management System EMS 5 10 Upgrading the Device s Software The procedure below describes how to upgrade the devices software i e cmp file using the EMS gt To upgrade the device s cmp file 1 From the Tools menu choose Software Manager the Software Manager screen appears Figure 5 11 Software Manager Screen W Software Manager File View Actions Help Managed Version VERSION 5 4 68 Managed Version VERSION 5 4 62 Managed Version VERSION 6 00A4L 006 002 SIP VERSION 2 Click the Add File i icon the Add Files dialog box appears Figure 5 12 Add Files Screen Add Files x SotmareFies MP M1K M2KAPM2K M3KAPM3K TP 260 Software CMP File Only CMP amp EMS amp INI Files CMP Software Version Major Version Select Product Select Protocol M5K M8KAPM5KAPMAK Software File Type File Name SW Description EMS INSTALL OK Cancel Version 6 0 205 March 2010 7a K tal AudioCodes MediaPack Series 3 Select the cmp file by performing the following a Ensure that the CMP File Only option is selected b Inthe CMP field click the browse button and navigate to the required cmp file the software version number of the selected file appears in the Software Version field c From the Major Version drop down list select the version number of the cmp file d From the Select Product drop dow
25. Rule 3 traffic from the subnet 10 31 4 xxx destined to ports 4000 9000 is always blocked regardless of protocol Rule 4 traffic from the subnet 10 4 xxx yyy destined to ports 4000 9000 is always blocked regardless of protocol All other traffic is allowed To edit a rule In the Edit Rule column select the rule that you want to edit Modify the fields as desired Click the Apply button to save the changes To save the changes to flash memory refer to Saving Configuration on page 161 To activate a de activated rule In the Edit Rule column select the de activated rule that you want to activate Click the Activate button the rule is activated To de activate an activated rule In the Edit Rule column select the activated rule that you want to de activate Click the DeActivate button the rule is de activated To delete a rule Select the radio button of the entry you want to activate Click the Delete Rule button the rule is deleted To save the changes to flash memory refer to Saving Configuration on page 161 71 March 2010 ca AudioCodes Table 3 9 Internal Firewall Parameters Parameter Is Rule Active Source IP AccessList Source IP Prefix Length AccessList PrefixLen Local Port Range AccessList Start Port AccessList End Port Protocol AccessList Protocol Packet Size AccessList Packet Size Byte Rate AccessList Byte Rate Burst Bytes AccessList Byte B
26. SIP User s Manual 6 Configuration Parameters Reference Parameter Description EMS Blind Transfer Disconnect Defines the duration in milliseconds for which the device Timeout waits for a disconnection from the Tel side after the Blind BlindTransferDisconnectTimeout Transfer Code KeyBlindTransfer has been identified When this timer expires a SIP REFER message is sent toward the IP side If this parameter is set to 0 the REFER message is immediately sent The valid range is 0 to 1 000 000 The default is 0 6 8 7 Three Way Conferencing Parameters The three way conferencing parameters are described in the table below Table 6 40 Three Way Conferencing Parameters Parameter Description Web Enable 3 Way Conference Enables or disables the 3 Way Conference feature EMS Enable 3 Way Disahi fault Enable3WayConference Meaple Disable eau 1 Enable Enables 3 way conferencing Note For this parameter to take effect a device reset is required Web 3 Way Conference Mode Defines the mode of operation when the 3 Way Conference EMS 3 Way Mode feature is used SWayConferenceMode 0 AudioCodes Media Server The Conference initiating INVITE sent by the device uses the ConferencelD concatenated with a unique identifier as the Request URI This same Request URI is set as the Refer To header value in the REFER messages that are sent to the two remote parties This conference mode is used when operating with
27. The displayed logged messages are color coded as follows e Yellow fatal error message e Blue recoverable error message i e non fatal error e Black notice message To clear the page of Syslog messages access the Message Log page again see Step 2 the page is cleared and new messages begin appearing To stop the Message Log Close the Message Log page by accessing any another page in the Web interface 173 March 2010 7a c tal AudioCodes MediaPack Series 3 5 1 2 Viewing Ethernet Port Information The Ethernet Port Information page displays read only information on the Ethernet connection used by the device This includes duplex mode and speed You can also access this page from the Home page refer to Using the Home Page on page 47 For detailed information on the Ethernet redundancy scheme refer to Ethernet Interface Redundancy For detailed information on the Ethernet interface configuration refer to Ethernet Interface Configuration on page 443 gt To view Ethernet port information m Open the Ethernet Port Information page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Ethernet Port Information page item Figure 3 111 Ethernet Port Information Page w Ethernet Information Port 1 Duplex Mode Port 1 Speed Table 3 30 Ethernet Port Information Parameters Parameter Description Port Duplex Mode Displays the Dupl
28. Web EMS Conference ID ConferencelD SIP User s Manual MediaPack Series Description Notes This parameter is applicable only to FXS interfaces When using an external conference server i e options 0 or 1 more than one three way conference may be supported up to six Currently the on board 3 way conference mode option 2 is not supported when using SRTP Determines the maximum number of simultaneous on board three way conference calls The valid range is 0 to 2 The default is 2 Notes For enabling on board three way conferencing use the parameter 3WayConferenceMode This parameter is applicable only to FXS interfaces Determines the ports that are not allocated as resources for on board three way conference calls that are initiated by other ports Ports that are not configured with this parameter and that are idle are used by the device as a resource for establishing these type of conference calls The valid range is up to 8 ports To add a range of ports use the comma separator For example for not allowing the use of ports 2 4 and 8 as resources enter the following value 2 4 8 The order of the entered values is not relevant i e the example above can be entered as 8 2 4 The default is 0 Notes To enable on board three way conferencing use the parameters 3WayConferenceMode and MaxInBoardConferenceCalls This parameter is applicable only to FXS interfaces Defi
29. c Right click the new entry and then select Unlock Rows d Click Apply and close the active window If a Proxy Server is not implemented map outgoing telephone calls to IP addresses Open the SIP Routing frame Configuration icon gt SIP Routing menu Select the Tel to IP tab a Click the al button to add a new entry and then click Yes to confirm the Tel to IP Routing table is displayed Double click each field to enter values Right click the new entry and select Unlock Rows d Click Apply and close the active window 197 March 2010 5 5 A e AudioCodes MediaPack Series Configuring Advanced IPSec IKE Parameters After you have pre configured IPSec via SSH refer to Securing EMS Device Communication on page 192 you can optionally configure additional IPSec and IKE entries for other SNMP Managers aside from the EMS Note Do not remove the default IPSec and IKE tables that were previously loaded to the device when you enabled IPSec To configure IPSec IKE tables In the MG Tree select the device Open the MG Info and Security Provisioning screen Configuration icon gt Info Security Frame menu Select the IPSec Proposal tab the IPSec Proposal screen is displayed Figure 5 6 IPSec Table Screen Parameters List 2 u IPSec Proposal General Info IPSec Enable Strict IKE certificate validation Disable F IPSec Proposal Qu a Web Access Addresses 70 2m m 2m
30. c tal AudioCodes MediaPack Series 9 6 4 V 152 Support The device supports the ITU T recommendation V 152 Procedures for Supporting Voice Band Data over IP Networks Voice band data VBD is the transport of modem facsimile and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals For V 152 capability the device supports T 38 as well as VBD codecs i e G 711 A law and G 711 u law The selection of capabilities is performed using the coders table refer to Configuring Coders on page 102 When in VBD mode for V 152 implementation support is negotiated between the device and the remote endpoint at the establishment of the call During this time initial exchange of call capabilities is exchanged in the outgoing SDP These capabilities include whether VBD is supported and associated RTP payload types gpmd SDP attribute supported codecs and packetization periods for all codec payload types ptime SDP attribute After this initial negotiation no Re INVITE messages are necessary as both endpoints are synchronized in terms of the other side s capabilities If negotiation fails i e no match was achieved for any of the transport capabilities fallback to existing logic occurs according to the parameter IsFaxUsed Below is an example of media descriptions of an SDP indicating support for V 152 i SS O O0 IN IPV4 lt IPAdressA gt 0 J c IN IP4 lt IPAddressA
31. case sensitive to the device If a message appears with the RSA host key click Yes to continue the shell prompt appears gt 2 At the CLI prompt type the command chpw and specify the existing and new passwords chpw lt old password gt lt new password gt where e lt old password gt is the existing password e lt new password gt is the new password The device responds with the message Password changed 3 Close the SSH client session and reconnect using the new password Note The default user name Admin cannot be changed from within an SSH client session Version 6 0 193 March 2010 V mI K AudioCodes MediaPack Series 5 3 Adding the Device in EMS Once you have defined the IPSec communication protocol for communicating between EMS and the device and configured the device s IP address refer to the device s Installation Manual you can add the device in the EMS Adding the device to the EMS includes the following main stages a Adding a Region b Defining the device s IP address and other initial settings gt To initially setup the device in EMS EMS 1 Start the EMS by double clicking the shortcut icon on your desktop or from the Start menu point to Programs point to EMS Client and then click EMS CLient the Login Screen appears Figure 5 2 EMS Login Screen Login Screen Version 6 0 48 SEE EM 6 0 AudioCodes ee oe 2 Enter your login user
32. 10 8 201 108 m F4200 OK 10 8 201 161 gt gt 10 8 201 108 SIP 2 0 200 OK Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 6000e10 8 201 108 gt tag 1c5354 To lt sip 2000 10 8 201 161 gt tag 1c7345 Call ID 534366556655skKw 6000 2000 10 8 201 108 CSeq 18153 INVITE Contact lt sip 2000 10 8 201 161 user phone gt Server Audiocodes Sip Gateway MediaPack v 6 00 010 006 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 206 o AudiocodesGW 30221 87035 IN IP4 10 8 201 161 s Phone Call CN IP O2 OHK ic 0 m audio 7210 RTP AVP 8 96 a rtpmap 8 pcma 8000 a ptime 20 a rtpmap 96 telephone event 8000 amo C iL5 SIP User s Manual 422 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities m F5 ACK 10 8 201 108 gt gt 10 8 201 10 ACK sip 2000 10 8 201 161 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacZYpJWxZ From lt sip 6000 10 8 201 108 gt tag 1c5354 To lt sip 2000 10 8 201 161 gt tag 1c7345 Call ID 534366556655skKw 6000 2000 10 8 201 108 User Agent Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeg 18153 ACK Supported 100rel em Content Length 0 Note Phone 6000 goes on hook and device 10 8 201 108 sends a BYE to device 10 8 201 161 A voice path is established m F6 BYE 10 8 201 108 gt gt
33. Click the Get Scenario File button the File Download window appears Click Save and then in the Save As window navigate to the folder to where you want to save the Scenario file When the file is successfully downloaded to your PC the Download Complete window appears Click Close to close the Download Complete window 3 1 8 5 Loading a Scenario to the Device Instead of creating a Scenario you can load a Scenario file data file from your PC to the device gt To load a Scenario to the device 1 On the Navigation bar click the Scenarios tab the Scenario appears in the Navigation tree 2 Click the Get Send Scenario File button located at the bottom of the Navigation tree the Scenario File page appears refer to Saving a Scenario to a PC on page 41 3 Click the Browse button and then navigate to the Scenario file stored on your PC 4 Click the Send File button You can only load a Scenario file to a device that has an identical hardware configuration setup to the device in which it was created For example if the Scenario was created in a device with FXS interfaces the Scenario cannot be loaded to a device that does not have FXS interfaces The loaded Scenario replaces any existing Scenario You can also load a Scenario file using BootP by loading an ini file that contains the ini file parameter ScenarioFileName refer to Web and Telnet Parameters on page 222 The Scenario dat file must be
34. LeaveFromRight number of digits to keep from the right side If both RemoveFromRight and LeaveFromRight are defined the RemoveFromRight is applied first The registered database contains the AoR before and after the manipulation The range of both RemoveFromRight and LeaveFromRight is 0 to 30 Note This table can include only one index entry This ini file table parameter configures the IP to IP Routing table for SAS routing rules The format of this parameter is as follows IP2IPRouting FORMAT IP2IPRouting Index IP2IPRouting_SrclPGroupID IP2IPRouting SrcUsernamePrefix IP2IPRouting_SrcHost IP2IPRouting DestUsernamePrefix IP2IPRouting DestHost IP2IPRouting_DestType IP2IPRouting_DestIPGroupID IP2IPRouting_DestSRDID IP2IPRouting_DestAddress IP2IPRouting DestPort IP2IPRouting_DestTransportType IP2IPRouting_AltRouteOptions IP2IPRouting For example IP2IPRouting 1 1 0 1 1 0 1 0 Notes This table can include up to 120 indices where 0 is the first index The parameters SrclPGroupID DestSRDID and AltRouteOptions are not applicable Fora detailed description of the individual parameters in this table and for configuring this table using the Web interface refer to Configuring the IP2IP Routing Table SAS on page 146 Fora description on configuring ini file table parameters refer to Configuring ini File Table Parameters on page 186 313 March 2010 ca AudioCo
35. TCP connections to all destinations are persistent and not released unless the device reaches 70 of its maximum TCP resources While trying to send a SIP message connection reuse policy determines whether alive connections to the specific destination are re used Persistent TCP connection ensures less network traffic due to fewer setting up and tearing down of TCP connections and reduced latency on subsequent requests due to avoidance of initial TCP handshake For TLS persistent connection may reduce the number of costly TLS handshakes to establish security associations in addition to the initial TCP connection set up Note If the destination is a Proxy server the TCP TLS connection is persistent regardless of the settings of this parameter Defines the Timer B INVITE transaction timeout timer and Timer F non INVITE transaction timeout timer as defined in RFC 3261 when the SIP Transport Type is TCP The valid range is 0 to 40 sec The default value is 64 SIPT1Rtx msec SIP destination port for sending initial SIP requests The valid range is 1 to 65534 The default port is 5060 Note SIP responses are sent to the port specified in the Via header 248 Document LTRT 65413 SIP User s Manual Parameter Web Use user phone in SIP URL EMS Is User Phone IsUserPhone Web Use user phone in From Header EMS Is User Phone In From IsUserPhonelnFrom Web Use Tel URI for Asserted Identity UseTelURIForAs
36. The called telephone number prefix The prefix can include up to 49 digits Note The prefix can be a single digit or a range of digits For available notations refer to Dialing Plan Notation for Routing and Manipulation on page 377 The calling telephone number prefix The prefix can include up to 49 digits Note The prefix can be a single digit or a range of digits For available notations refer to Dialing Plan Notation for Routing and Manipulation on page 377 132 Document LTRT 65413 SIP User s Manual Parameter Source IP Address 3 Web Based Management Description The source IP address of an IP to Tel call obtained from the Contact header in the INVITE message that can be used for routing decisions Notes You can configure from where the source IP address is obtained using the parameter SourcelPAddressInput The source IP address can include the following wildcards v x depicts single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 and 10 8 8 99 v depicts any number between 0 and 255 For example 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 Calls matching all or any combination of the above characteristics are sent to the Hunt Group ID defined below Note For alternative routing additional entries of the same characteristics can be configured Hunt Group ID IP Profile ID Source IP Group ID Version 6 0 The Hunt Grou
37. The device reverses the polarity of the endpoint marking it unusable relevant for example for PBX DID lines This option can t be configured on the fly 8 Reorder Tone Polarity Reversal Same as 2 and 3 combined This option can t be configured on the fly 4 Current Disconnect The device disconnects the current of the FXS endpoint This option can t be configured on the fly Note This parameter is applicable only to FXS interfaces The time interval in msec between the first transmission of a SIP message and the first retransmission of the same message The default is 500 Note The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx For example assuming that SipT1Rtx 500 and SipT2Rtx 4000 The first retransmission is sent after 500 msec The second retransmission is sent after 1000 2 500 msec The third retransmission is sent after 2000 2 1000 msec The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent after 4000 2 2000 msec 262 Document LTRT 65413 SIP User s Manual Parameter Web SIP T2 Retransmission Timer msec EMS T2 RTX SipT2Rtx Web SIP Maximum RTX EMS Max RTX SIPMaxRtx Web Number of RTX Before Hot Swap EMS Proxy Hot Swap Rtx HotSwapRtx 6 7 2 6 Configuration Parameters Reference Description The maximum interval in msec between
38. VolPerfectHD What s Inside Matters Your Gateway To VoIP and 3GX are trademarks or registered trademarks of AudioCodes Limited All other products or trademarks are property of their respective owners Product specifications are subject to change without notice WEEE EU Directive Pursuant to the WEEE EU Directive electronic and electrical waste must not be disposed of with unsorted waste Please contact your local recycling authority for disposal of this product Customer Support Customer technical support and service are provided by AudioCodes Distributors Partners and Resellers from whom the product was purchased For Customer support for products purchased directly from AudioCodes contact support audiocodes com Abbreviations and Terminology Each abbreviation unless widely used is spelled out in full when first used Only industry standard terms are used throughout this manual Hexadecimal notation is indicated by Ox preceding the number Regulatory Information The Regulatory Information can be viewed at http www audiocodes com downloads Version 6 0 15 March 2010 E tall AudioCodes MediaPack Series Related Documentation Manual Name Product Reference Manual SIP CPE Devices MP 11x amp MP 124 SIP Release Notes MP 11x amp MP 124 SIP Installation Manual MP 11x SIP Fast Track Guide MP 124 SIP Fast Track Guide CPE Configuration Guide for IP Voice Mail Warning The device is supplied as a sealed un
39. gt Phone PSTN S gt Phone Fe FAX MediaPack MediaPack FXO FXS Router MediaPack oe 1 2 MediaPack Features This section provides a high level overview of some of the many device supported features For more updated information on the device s supported features refer to the latest MP 11x amp MP 124 SIP Release Notes 1 2 1 MP 11x Hardware Features The MP 11x series hardware features include the following m Combined FXS FXO devices four FXS and four FXO ports on the MP 118 two FXS and two FXO ports on the MP 114 MP 11x compact rugged enclosure only one half of a 19 inch rack unit 1 U high Lifeline provides a wired phone connection to the PSTN line that becomes active upon a power or network failure combined FXS FXO devices provide a Lifeline connection that s available on all FXS ports m LEDs on the front panel that provide information on the device s operating status and the network interface m Reset button on the rear panel for restarting the MP 11x and for restoring the MP 11x parameters to their factory default settings SIP User s Manual 18 Document LTRT 65413 SIP User s Manual 1 Overview 1 2 2 MP 124 Hardware Features The MP 124 hardware features include the following m MP 124 19 inch 1U rugged enclosure provides up to 24 analog FXS ports using a single 50 pin Telco connector m LEDs on the front panel that provide information on the device s operating
40. refer to Configuring the Proxy Sets Table on page 97 If you are not using a Proxy server you must configure the Tel to IP Routing described in Configuring the Tel to IP Routing on page 126 Defines the Home Proxy domain name If specified this name is used as the Request URI in REGISTER INVITE and other SIP messages and as the host part of the To header in INVITE messages If not specified the Proxy IP address is used instead The value must be string of up to 49 characters 265 March 2010 ca AudioCodes Parameter Web Redundancy Mode EMS Proxy Redundancy Mode ProxyRedundancyMode Web Proxy IP List Refresh Time EMS IP List Refresh Time ProxyIPListRefreshTime Web Enable Fallback to Routing Table EMS Fallback Used IsFallbackUsed Web EMS Prefer Routing Table PreferRouteTable Web EMS Always Use Proxy AlwaysSendToProxy Web SIP ReRouting Mode EMS SIP Re Routing Mode SIPReroutingMode SIP User s Manual MediaPack Series Description Determines whether the device switches back to the primary Proxy after using a redundant Proxy 0 Parking device continues working with a redundant now active Proxy until the next failure after which it works with the next redundant Proxy default 1 Homing device always tries to work with the primary Proxy server i e switches back to the primary Proxy whenever it s available Note To use this Proxy Redundancy mechanism
41. 16 when the call duration is zero Indicates the number of calls that were terminated due to a call forward The counter is incremented as a result of the following release reason RELEASE BECAUSE FORWARD Indicates the number of calls whose destinations weren t found It is incremented as a result of one of the following release reasons GWAPP_UNASSIGNED_NUMBER 1 GWAPP_NO_ROUTE_TO_DESTINATION 3 Indicates the number of calls that failed due to mismatched device capabilities It is incremented as a result of an internal identification of capability mismatch This mismatch is reflected to CDR via the value of the parameter DefaultReleaseReason default is GWAPP_NO_ROUTE_TO_DESTINATION 3 or by the 179 March 2010 ca AudioCodes Counter Number of Failed Calls due to No Resources Number of Failed Calls due to Other Failures Average Call Duration ACD sec Attempted Fax Calls Counter Successful Fax Calls Counter MediaPack Series Description GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED 79 reason Indicates the number of calls that failed due to unavailable resources or a device lock The counter is incremented as a result of one of the following release reasons GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED RELEASE_BECAUSE_GW_LOCKED This counter is incremented as a result of calls that failed due to reasons not covered by the other counters The average call duration ACD in seconds of establishe
42. A unique accounting identifier match start amp stop For how many seconds the user received the service 437 Value Format String Numeric String String String String String String Numeric Numeric Numeric Numeric String Numeric 9 IP Telephony Capabilities Example Stop Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc SIPIDString abcde ac com Stop Yes No Age Start 8004567145 A CC Stop Acc Start Acc Stop Acc 2427456425 5135672127 Start Acc Stop Acc 1 start 2 stop Start Acc Stop Acc Stop Acc Stop Acc Start Acc Stop Acc 34832 Stop Acc March 2010 ca AudioCodes Attribute Number Attribute Response Attributes 26 44 VSA Value MediaPack Series 1 Name No Purpose Format Example AAA Number of packets Nene Stop received during the call Acc Number of packets sent N Stop umeric during the call Acc Physical port type of i Start i 0 Acc device on which the callis String f Asynchronous Stop active Acc H323 The reason for failing Return 103 authentication 0 ok Numeric ie seg rie Code other number failed p A unique accounting peer identifier match start amp String ap Session ID Acc stop Below is an example of RADIUS Accounting where the non standard parameters are preceded with brackets Accounting Request 361 Usercnane W acct session
43. Both AltRoutingTel2IPMode 2 or 3 This parameter is reset if no QoS information is received for 2 minutes Displays QoS information delay and packet loss calculated according to previous calls Notes This parameter is applicable only if the parameter Alt Routing Tel to IP Mode is set to QoS or Both AltRoutingTel2IPMode 2 or 3 This parameter is reset if no QoS information is received for 2 minutes DNS status can be one of the following DNS Disable DNS Resolved DNS Unresolved 184 Document LTRT 65413 SIP User s Manual 4 INI File Configuration 4 INI File Configuration The device can also be configured by loading an ini file containing user defined parameters The ini file can be loaded to the device using the following methods m Web interface refer to Backing Up and Restoring Configuration on page 171 m AudioCodes BootP TFTP utility refer to the Product Reference Manual m Any standard TFTP server The ini file configuration parameters are saved in the device s non volatile memory when the file is loaded to the device If a parameter is excluded from the loaded ini file the default value is assigned to that parameter according to the cmp file running on the device thereby overriding the value previously defined for that parameter For a list and description of the ini file parameters refer to Configuration Parameters Reference on page 207 Some parameters are configurabl
44. Configuration Parameters Reference Description lower channel 5 Dest Number Cyclic Ascending The device first selects the channel according to the called number If the called number isn t found it then selects the next available channel in ascending cyclic order Note that if the called number is found but the port associated with this number is busy the call is released 6 By Source Phone Number The device selects the channel according to the calling number Notes For defining the channel select mode per Hunt Group refer to Configuring Hunt Group Settings on page 85 The phone numbers of the device s channels are defined by the TrunkGroup parameter Defines the default destination phone number which is used if the received message doesn t contain a called party number and no phone number is configured in the Endpoint Phone Number Table refer to Configuring the Endpoint Phone Numbers on page 143 This parameter is used as a starting number for the list of channels comprising all the device s Hunt Groups The default value is 1000 Determines the IP address that the device uses to determine the source of incoming INVITE messages for IP to Tel routing 1 Not configured default 0 SIP Contact Header The IP address in the Contact header of the incoming INVITE message is used 1 Layer 3 Source IP The actual IP address Layer 3 from where the SIP packet was received is used
45. Configuring Coder Groups on page 104 or the device s default coder refer to Configuring Coders on page 102 to which you want to assign the Profile Repeat steps 2 through 6 to configure additional Tel Profiles optional Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 5 4 Configuring IP Profiles The IP Profile Settings page allows you to define up to nine different IP Profiles You can later assign IP Profiles to routing rules in the call routing tables Tel to IP Routing page refer to Configuring Tel to IP Routing on page 126 IP to Hunt Group Routing Table page refer to Configuring the IP to Hunt Group Routing Table on page 131 The IP Profile Settings page conveniently groups the different parameters according to application to which they pertain Version 6 0 Common Parameters parameters common to all application types Gateway Parameters parameters applicable to gateway functionality For a detailed description of each parameter in the IP Profile table refer to its corresponding global parameter configured as an individual parameter IP Profiles can also be implemented when operating with a Proxy server when the parameter AlwaysUseRouteTable is set to 1 You can also configure the IP Profiles using the ini file table parameter IPProfile refer to SIP Configuration Parameters on page 245
46. EnableSyslog SIP User s Manual Description IP address in dotted decimal notation of the computer you are using to run the Syslog server The Syslog server is an application designed to collect the logs and error messages generated by the device Default IP address is 0 0 0 0 For information on Syslog refer to the Product Reference Manual Defines the UDP port of the Syslog server The valid range is 0 to 65 535 The default port is 514 For information on the Syslog refer to the Product Reference Manual Sends the logs and error message generated by the device to the Syslog server 0 Disable Logs and errors are not sent to the Syslog server default 1 Enable Enables the Syslog server Notes For this parameter to take effect a device reset is required If you enable Syslog you must enter an IP address and a port 226 Document LTRT 65413 SIP User s Manual Parameter SyslogOutputMethod MaxBundleSyslogLength Web CDR Server IP Address EMS IP Address of CDR Server CDRSyslogServerlP Web EMS CDR Report Level CDRReportLevel Version 6 0 6 Configuration Parameters Reference Description number using the SyslogServerlP and SyslogServerPort parameters You can configure the device to send Syslog messages implementing Debug Recording by using the SyslogOutputMethod parameter For a detailed description on Debug Recording refer to the Product Reference Manual
47. In this scenario multiple network interface capabilities are not available Defines a string up to 16 characters to name this interface This name is displayed in management interfaces Web CLI and SNMP for better readability and has no functional use as well as the SIP Media Realm table refer to Configuring Media Realms Note The interface name is a mandatory parameter and must be unique for each interface For a description of this parameter refer to Networking Parameters on page 207 For a description of this parameter refer to Networking Parameters on page 207 3 3 1 2 Configuring the Application Settings The Application Settings page is used for configuring various application parameters such as Network Time Protocol NTP daylight saving time and Telnet For a description of these parameters refer to Configuration Parameters Reference on page 207 SIP User s Manual 54 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To configure the Application settings 1 Open the Application Settings page Configuration tab gt Network Settings menu gt Application Settings page item Figure 3 36 Application Settings Page w NTP Settings NTP Server IP Address 0 0 0 0 NTP UTC Offset Hours fos Minutes 0 NTP Updated Interval Hours Minutes 0 w Day Light Saving Time Day Light Saving Time Disable v Start Time m OE End Time Je ny fo Jo Of
48. Indication on page 416 Enables the visual display of MWI 0 Disable Disable default 1 Enable Enables visual MWI by supplying line voltage of approximately 100 VDC to activate the phone s lamp Note This parameter is applicable only for FXS interfaces Determines whether MWI information is sent to the phone display 0 Disable MWI information isn t sent to display default 1 Enable The device generates an MWI message determined by the parameter CallerlDType which is displayed on the MWI display Note This parameter is applicable only to FXS interfaces Enables subscription to an MWI server 0 No Disables MWI subscription default 1 Yes Enables subscription to an MWI server defined by the parameter MWIServerlP address Note To configure whether the device subscribes per endpoint or per the entire device use the parameter SubscriptionMode MWI server s IP address If provided the device subscribes to this IP address The MWI server address can be configured as a numerical IP address or as a domain name If not configured the Proxy IP address is used instead 299 March 2010 ca AudioCodes Parameter Web EMS MWI Server Transport Type MWIServerTransportType Web MWI Subscribe Expiration Time EMS MWI Expiration Time MWIExpirationTime Web MWI Subscribe Retry Time EMS Subscribe Retry Time SubscribeRetry Time Web Subscription Mode SubscriptionMode
49. MP 114 8 MP 118 42 x 172 x 220 mm MP 124 44 x 445 x 269 mm Rack mount Table top Wall mount Applying 100V DC online for lighting bulb in handset FSK Stutter Dial Tone PSTN Fallback Support of PSTN fallback due to Power failure if the IP connection is down or due to customer defined IP QoS thresholds Stand Alone Survivability SAS Supports SAS of up to 25 SIP users UA Sine 54 V RMS typical balanced ringing only 25 100Hz Ringer Equivalency Number REN 3 Up to 1500 ohm for the MP 11x Up to 1600 ohm for the MP 124 468 Document LTRT 65413 SIP User s Manual Function Lifeline Caller ID Polarity Reversal Wink Metering Tones Distinctive Ringing Message Waiting Indication Outdoor Protection Homologation EMC Safety Telecom Version 6 0 12 Selected Technical Specifications Specification Supported in all ports of Mixed FXS FXO and in first port of MP 114 FXS and MP 118 FXS using special Lifeline cable Bellcore GR 30 CORE Type 1 using Bell 202 FSK modulation ETSI Type 1 NTT Denmark India Brazil British and DTMF ETSI CID ETS 300 659 1 Immediate or smooth to prevent erroneous ringing 12 16 KHz sinusoidal bursts Generation on FXS By frequency 15 100 Hz and cadence patterns DC voltage generation TIA EIA 464 B V23 FSK data Stutter dial tone Over voltage protection and surge immunity Note Supported only on MP 124D EN55022 Class B CFR Part 15 Class B EN5502
50. PREFIX_Profileld PREFIX MeteringCode PREFIX_DestPort PREFIX_SrclPGroupID PREFIX_DestHostPrefix PREFIX_DestIPGroupID PREFIX_SrcHostPrefix PREFIX_TransportType PREFIX_SrcTrunkGroupID PREFIX For example PREFIX 0 guest 0 255 1 1 1 1 PREFIX 1 20 10 33 37 77 0 255 1 2 0 1 PREFIX 2 30 10 33 37 79 1 255 1 1 2 1 Notes This parameter can include up to 50 indices 334 Document LTRT 65413 SIP User s Manual Parameter 6 Configuration Parameters Reference Description Fora detailed description of the table s parameters and for configuring this table using the Web interface refer to Configuring the Tel to IP Routing on page 126 The parameters PREFIX_SrclPGroupID PREFIX_DestHostPrefix and PREFIX_SrcHostPrefix are not applicable Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web IP to Hunt Group Routing Table EMS SIP Routing gt IP to Hunt PSTNPrefix Version 6 0 This ini file table parameter configures the routing of IP calls to Hunt Groups The format of this parameter is as follows PSTNPrefix FORMAT PstnPrefix Index PstnPrefix DestPrefix PstnPrefix TrunkGroupld PstnPrefix SourcePrefix PstnPrefix SourceAddress PstnPrefix Profileld PstnPrefix_SrclPGroupID PstnPrefix DestHostPrefix PstnPrefix SrcHostPrefix PSTNPrefix For example Ps
51. STUN Server Primary IP EMS Primary Server IP STUNServerPrimaryIP Web STUN Server Secondary IP EMS Secondary Server IP STUNServerSecondaryIP STUNServerDomainName NAT Parameters EMS Binding Life Time NATBindingDefaultTimeout Web NAT IP Address EMS Static NAT IP Address StaticNatIP EMS Disable NAT DisableNAT Version 6 0 6 Configuration Parameters Reference Description Defines the IP address of the primary STUN server The valid range is the legal IP addresses The default value is 0 0 0 0 Note For this parameter to take effect a device reset is required Defines the IP address of the secondary STUN server The valid range is the legal IP addresses The default value is 0 0 0 0 Note For this parameter to take effect a device reset is required Defines the domain name for the Simple Traversal of User Datagram Protocol STUN server s address used for retrieving all STUN servers with an SRV query The STUN client can perform the required SRV query to resolve this domain name to an IP address and port sort the server list and use the servers according to the sorted list Notes For this parameter to take effect a device reset is required Use either the STUNServerPrimarylP or the STUNServerDomainName parameter with priority to the first one Defines the default NAT binding lifetime in seconds STUN refreshes the binding information after this time expires The valid ran
52. Silence Coder Name Packetization Time Payload Type 5 Suppression 30 v IIs Disabled v 2 From the Coder Group ID drop down list select a coder group ID SIP User s Manual 104 Document LTRT 65413 SIP User s Manual 3 Web Based Management 10 11 Version 6 0 From the Coder Name drop down list select the first coder for the coder group From the Packetization Time drop down list select the packetization time in msec for the coder The packetization time determines how many coder payloads are combined into a single RTP packet From the Rate drop down list select the bit rate in kbps for the coder you selected In the Payload Type field if the payload type i e format of the RTP payload for the coder you selected is dynamic enter a value from 0 to 120 payload types of well known coders cannot be modified From the Silence Suppression drop down list enable or disable the silence suppression option for the coder you selected Repeat steps 3 through 7 for the next coders optional Repeat steps 2 through 8 for the next coder group optional Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 105 March 2010 3 3 4 5 3 Configuring Tel Profile ca AudioCodes MediaPack Series The Tel Profile Settings page allows you to define up to nine Tel Profiles You can then assign the
53. These Coder Groups can later be assigned to IP or Tel Profiles The format of this parameter is as follows CodersGroup0 FORMAT CodersGroup0 Index CodersGroup0 Name CodersGroup0 pTime CodersGroupO rate CodersGroup0_PayloadType CodersGroup0 Sce CodersGroup0O 284 Document LTRT 65413 SIP User s Manual Parameter Version 6 0 6 Configuration Parameters Reference Description Where Index Coder entry 0 9 i e up to 10 coders per group Name Coder name Ptime Packetization time ptime how many coder payloads are combined into a single RTP packet Rate Packetization rate PayloadType Identifies the format of the RTP payload Sce Enables silence suppression y 0 Disabled default v 1 Enabled For example below are defined two Coder Groups 0 and 1 CodersGroup0 FORMAT CodersGroupO Index CodersGroup0O Name CodersGroup0O pTime CodersGroup0 rate CodersGroup0 PayloadType CodersGroup0 Sce CodersGroup0O 0 g711Alaw64k 20 0 255 0 CodersGroupO 1 eg711Ulaw 10 0 71 0 CodersGroup0 2 eg711Ulaw 10 0 71 0 NXCodersGroup0 ll CodersGroupl FORMAT CodersGroupl Index CodersGroupl Name CodersGroupl pTime CodersGroupl rate CodersGroupl PayloadType CodersGroupl Sce CodersGroupl 0 Transparent 20 0 56 0 CodersGroupl 1 9726 20 0 23 0 CodersGroup1 The table below lists the supported coders Coder Name Packetizat
54. This menu includes the following page items m Load Auxiliary Files refer to Loading Auxiliary Files on page 163 m Software Upgrade Key refer to Loading a Software Upgrade Key on page 165 m Software Upgrade Wizard refer to Software Upgrade Wizard on page 168 E Configuration File refer to Backing Up and Restoring Configuration on page 171 3 4 2 1 Loading Auxiliary Files The Load Auxiliary Files page allows you to load various auxiliary files to the device These auxiliary files are briefly described in the table below Table 3 29 Auxiliary Files Descriptions File Type Description ini Provisions the device s parameters The Web interface enables practically full device provisioning but customers may occasionally require new feature configuration parameters in which case this file is loaded Note Loading this file only provisions those parameters that are included in the ini file Parameters that are not specified in the ini file are reset to factory default values Call Progress This is a region specific telephone exchange dependent file that contains the Tones Call Progress Tones CPT levels and frequencies that the device uses The default CPT file is U S A Prerecorded The dat PRT file enhances the device s capabilities of playing a wide range of Tones telephone exchange tones that cannot be defined in the Call Progress Tones file Dial Plan Dial plan file User Info The User Information file maps PBX exte
55. This results in the best packet error performance but at the cost of extra delay At the minimum value of 0 the buffer tracks delays only to compensate for clock drift and quickly decays back to the minimum level This optimizes the delay performance but at the expense of a higher error rate The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide a good compromise between delay and error rate The jitter buffer holds incoming packets for 10 msec before making them available for decoding into voice The coder polls frames from the buffer at regular intervals in order to produce continuous speech As long as delays in the network do not change jitter by more than 10 msec from one packet to the next there is always a sample in the buffer for the coder to use If there is more than 10 msec of delay at any time during the call the packet arrives too late The coder tries to access a frame and is not able to find one The coder must produce a voice sample even if a frame is not available It therefore compensates for the missing packet by adding a Bad Frame Interpolation BFI packet This loss is then flagged as the buffer being too small The dynamic algorithm then causes the size of the buffer to increase for the next voice session The size of the buffer may decrease again if the device notices that the buffer is not filling up as much as expected At no time does the buffer decrease to less than the minimum siz
56. field in the Source Number Manipulation table refer to Configuring the Number Manipulation Tables on page 115 You can also configure the Caller Display Information table using the ini file table parameter CallerDisplayInfo 139 March 2010 7a tal AudioCodes MediaPack Series 3 3 4 9 4 Configuring Call Forward The Call Forwarding Table page allows you to forward redirect IP to Tel calls using SIP 302 response originally destined to specific device ports to other device ports or to an IP destination Ensure that the Call Forward feature is enabled default for the settings on this page to take effect To enable Call Forward use the parameter EnableForward Configuring Supplementary Services on page 111 You can also configure the Call Forward table using the ini file table parameter Fwdlnfo gt To configure Call Forward per port 1 Open the Call Forward Table page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Call Forward page item Figure 3 87 Call Forward Table Page Forward to Phone Time for No Reply Number Forward Gateway Port Port1 FXS On busy z 201 30 Forward Type Port 2 FXS On busy mj 201 30 Port 3 FXS No Answer 203 Port4 FXS Unconditional 202 10 2 1 1 Port 5 FXO Deactivate 2 Configure the Call Forward parameters for each port according to the
57. must use the same pre shared key for the authentication process to succeed Notes This parameter is applicable only if the Authentication Method parameter is set to pre shared key The pre shared key forms the basis of IPSec security and therefore it should be handled with care the same as sensitive passwords It is not recommended to use the same pre shared key for several connections Since the ini file is plain text loading it to the device over a secure network connection is recommended Use a secure transport such as HTTPS or a direct crossed cable connection from a management PC After it is configured the value of the pre shared key cannot be retrieved Defines the source port to which this configuration applies The default value is 0 i e any port Defines the destination port to which this configuration applies The default value is 0 i e any port 81 March 2010 ca AudioCodes Parameter Name Protocol IPsecSATable Protocol IKE SA Lifetime IPsecSATable Phase1SaLifetimeln Sec IPSec SA Lifetime sec IPsecSATable Phase2SaLifetimeln Sec IPSec SA Lifetime Kbs IPsecSATable Phase2SaLifetimeln KB Dead Peer Detection Mode IPsecSATable DPDmode Remote Tunnel Addr IPsecSATable RemoteTunnelAddre ss Remote Subnet Addr IPsecSATable_RemoteSubnetIPAdd ress SIP User s Manual MediaPack Series Description Defines the protocol type to which this configuration a
58. on page 186 Method for allocating incoming IP to Tel calls to a channel port 0 By Dest Phone Number Selects the device s channel according to the called number default 1 Cyclic Ascending Selects the next available channel in an ascending cyclic order Always selects the next higher channel number in the Hunt Group When the device reaches the highest channel number in the Hunt Group it selects the lowest channel number in the Hunt Group and then starts ascending again 2 Ascending Selects the lowest available channel It always starts at the lowest channel number in the Hunt Group and if that channel is unavailable selects the next higher channel 3 Cyclic Descending Selects the next available channel in descending cyclic order It always selects the next lower channel number in the Hunt Group When the device reaches the lowest channel number in the Hunt Group it selects the highest channel number in the Hunt Group and then starts descending again 4 Descending Selects the highest available channel It always starts at the highest channel number in the Hunt Group and if that channel is unavailable selects the next 332 Document LTRT 65413 SIP User s Manual Parameter Web Default Destination Number DefaultNumber Web Source IP Address Input SourcelPAddressInput Web Use Source Number As Display Name EMS Display Name UseSourceNumberAsDisplayN ame Version 6 0 6
59. on page 457 for more details m Apart from the interface having the default gateway defined the Gateway column for all other interfaces must be set to 0 0 0 0 for IPv4 m The Interface Name column may have up to 16 characters This column allows the user to name each interface with an easier name to associate the interface with This column must have a unique value to each interface and must not be left blank m For IPv4 interfaces the Interface Mode column must be set to IPv4 Manual numeric value 10 m When defining more than one interface of the same address family VLANs must be enabled the VlanMode should be set to 1 m VLANs become available only when booting the device from flash When booting using BootP DHCP protocols VLANs are disabled to allow easier maintenance access In this scenario multiple network interface capabilities are not available m The Native VLAN ID may be defined using the VianNativeVlanld parameter This relates untagged incoming traffic as if reached with a specified VLAN ID Outgoing traffic from the interface which VLAN ID equals to the Native VLAN ID are tagged with VLAN ID 0 priority tag m Quality of Service parameters specify the priority field for the VLAN tag IEEE 802 1p and the DiffServ field for the IP headers These specifications relate to service classes m When booting using BootP DHCP protocols the address received from the BootP DHCP server acts as a temporary OAMP add
60. only The Caller ID interworking can be changed using the parameters UseSourceNumberAsDisplayName and UseDisplayNameAsSourceNumber 9 7 8 2 Debugging a Caller ID Detection on FXO The procedure below describes debugging caller ID detection in FXO interfaces gt 1 To debug a Caller ID detection on an FXO interface Verify that the parameter EnableCallerlD is set to 1 Verify that the caller ID standard and substandard of the device matches the standard of the PBX using the parameters CallerlDType BellcoreCallerlIDTypeOneSubStandard and ETSICallerIDTypeOneSubStandard Define the number of rings before the device starts the detection of caller ID using the parameter RingsBeforeCallerlD Verify that the correct FXO coefficient type is selected using the parameter CountryCoefficients as the device is unable to recognize caller ID signals that are distorted Connect a phone to the analog line of the PBX instead of to the device s FXO interface and verify that it displays the caller ID If the above does not solve the problem you need to record the caller ID signal and send it to AudioCodes as described below Version 6 0 417 March 2010 Aa c tal AudioCodes MediaPack Series 9 7 8 3 To record the caller ID signal using the debug recording mechanism 1 Access the FAE page by appending FAE to the device s IP address in the Web browser s URL for example http 10 13 4 13 FAE 2 Press the C
61. page as described in Configuring the Management Settings on page 152 2 In the SNMP Trap Destinations field click the right pointing arrow button the SNMP Trap Destinations page appears Figure 3 96 SNMP Trap Destinations Page IP Address Trap Enable SNMP Manager 10 8 2 28 Enable W m SNMP Manager 0 0 0 0 Enable w d SNMP Manager 0 0 0 0 Enable m SNMP Manager 0 0 0 0 Enable m SNMP Manager 0 0 0 0 Enable Configure the SNMP trap managers parameters according to the table below Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 Note Only table row entries whose corresponding check boxes are selected are applied when clicking Submit otherwise settings revert to their defaults Table 3 26 SNMP Trap Destinations Parameters Description Parameter Description SNMP Manager Determines the validity of the parameters IP address and SNMPManagerlsUsed x port number of the corresponding SNMP Manager used to receive SNMP traps 0 Check box cleared Disabled default 1 Check box selected Enabled IP Address IP address of the remote host used as an SNMP SNMPManagerTablelP x Manager The device sends SNMP traps to these IP addresses Enter the IP address in dotted decimal notation e g 108 10 1 255 SIP User s Manual 154 Document LTRT 65413 SIP User
62. please contact the AudioCodes Distributor and Reseller from whom this product was purchased 465 March 2010 A c tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 466 Document LTRT 65413 SIP User s Manual 12 Selected Technical Specifications 12 Selected Technical Specifications The main technical specifications of the MP 11x and MP 124 devices are listed in the table below Note Function Interfaces Voice Ports Telephone Interfaces Lifeline Network Interface Indicators Channel Voice Fax Modem Voice over Packet Capabilities Voice Compression Fax over IP 3 Way Conference QoS IP Transport Signaling Signaling Version 6 0 All specifications in this document are subject to change without prior notice Table 12 1 MediaPack Technical Specifications Specification MP 112 2 ports MP 114 4 ports MP 118 8 ports MP 124 24 ports MP 112 FXS RJ 11 MP 114 8 MP 118 FXS FXO or mixed FXS FXO RJ 11 MP 124 FXS 50 pin Telco Automatic cut through of a single analog line FXS version only refers only for the middle column 4 8 ports 10 100Base TX RJ 45 Status and activity LEDs G 168 2004 compliant Echo Cancellation VAD CNG Dynamic programmable Jitter Buffer modem detection and auto switch to PCM G 711 G 723 1 G 726 G 729A EG 711 G 722 T 38 compliant Group 3 fax relay up to 14 4 kbps with automatic switching to P
63. reminder ring xml body The NOTIFY request is sent from the Application Server to the device each time the Application Server forwards an incoming call The service is cancelled when an UNSUBSCRIBE request is sent from the device or when the Subscription time expires The Reminder Ring tone can be defined by using the parameter CallForwardRingTonelD which points to a ring tone defined in the Call Progress Tone file The following parameters are used to configure this feature m EnableNRTSubscription m ASSubscribelPGroupID m NRTRetrySubscriptionTime m CallForwardRingTonelD Call Forward Reminder Off Hook Special Dial Tone The device plays a special dial tone Stutter Dial tone Tone Type 15 to a specific FXS endpoint when the phone is off hooked and when a third party Application server AS e g a softswitch is used to forward calls intended for the endpoint to another destination This is useful in that it reminds the FXS user of this service This feature does not involve device subscription SIP SUBSCRIBE to the AS Activation deactivation of the service is notified by the server An unsolicited SIP NOTIFY request is sent from the AS to the device when the Call Forward service is activated or cancelled Depending on this NOTIFY request the device plays either the standard dial tone or the special dial tone for Call Forward SIP User s Manual 414 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities F
64. then the Request URI host name in the INVITE message is set to the value defined for the parameter Dest IP Address above otherwise if no IP address is defined it is set to the value of the parameter SIP Group Name defined in the IP Group table This parameter is used as the Serving IP Group in the Account table for acauiring authentication user password for this call For defining Proxy Set ID s refer to Configuring the Proxy Sets Table on page 97 IP Profile ID defined by the parameter IPProfile assigned to this IP destination call This allows you to assign numerous configuration attributes e g voice codes per routing rule Read only field displaying the Quality of Service of the destination IP address p a Alternative Routing feature is disabled OK IP route is available Ping Error No ping to IP destination route is unavailable QoS Low Poor QoS of IP destination route is unavailable DNS Error No DNS resolution only when domain name is used instead of an IP address Optional Charge Code 1 to 25 assigned to the routing rule For configuring Charge Codes refer to Configuring the Charge Codes Table on page 113 Note This parameter is applicable only to FXS interfaces 130 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 8 4 Configuring the IP to Hunt Group Routing Table The IP to Hunt Group Routing Table page allows you to configure up to
65. 0 202 202 0 0 RoutingTableDestinationPrefixLensColumn 16 16 RoutingTableGatewaysColumn 192 168 0 2 192 168 0 3 RoutingTableInterfacesColumn 0 0 RoutingTableHopsCountColumn 1 1 Version 6 0 461 March 2010 7a tal AudioCodes MediaPack Series Example 2 Three Interfaces one for each application exclusively the Multiple Interface table is configured with three interfaces one exclusively for each application type one interface for OAMP applications one for Call Control applications and one for RTP Media applications Table 10 11 Multiple Interface Table Example 2 Index Application Interface IP Address TUIR Peet wich FEMS Length Gateway ID Name 0 OAMP IPv4 192 168 85 14 16 0 0 0 0 1 ManagementlF 1 Control IPv4 200 200 85 14 24 0 0 0 0 200 myControllF 2 Media IPv4 211 211 85 14 24 211 211 85 1 211 myMedialF VLANs are required The Native VLAN ID is the same VLAN ID as the Management interface Index 0 One routing rule is required to allow remote management from a host in 176 85 49 0 24 Table 10 12 Routing Table Example 2 Destination Prefix Length Subnet Mask Gateway Interface Metric 176 85 49 0 24 192 168 0 1 0 1 All other parameters are set to their respective default values The ini file matching this configuration can be written as follows Interface Table Configuration InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes InterfaceTable Interfa
66. 10 8 201 10 BYE sip 2000 10 8 201 161 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacRKCVBud From lt sip 6000 10 8 201 108 gt tag 1c5354 To lt sip 2000 10 8 201 161 gt tag 1c7345 Call ID 534366556655skKw 6000 2000 10 8 201 108 User Agent Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeq 18154 BYE Supported 100rel em Content Length 0 m F7 OK 200 10 8 201 10 gt gt 10 8 201 108 SIP 2 0 200 OK Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacRKCVBud From lt sip 6000 10 8 201 108 gt tag 1c5354 To lt sip 2000 10 8 201 161 gt tag 1c7345 Call ID 534366556655skKw 6000 2000 10 8 201 108 Server Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeg 18154 BYE Supported 100rel em Content Length 0 9 8 2 SIP Authentication Example The device supports basic and digest MD5 authentication types according to SIP RFC 3261 standard A proxy server might require authentication before forwarding an INVITE message A Registrar Proxy server may also require authentication for client registration A proxy replies to an unauthenticated INVITE with a 407 Proxy Authorization Required response containing a Proxy Authenticate header with the form of the challenge After sending an ACK for the 407 the user agent can then re send the INVITE with a Proxy Authorization header containing the credentials User agents Redirect or Registrar servers typically use 401 Unauthorized response
67. 2010 7a e AudioCodes MediaPack Series Each call can be associated with one or two Profiles Tel Profile and or IP Profile If both IP and Tel profiles apply to the same call the coders and other common parameters of the preferred Profile determined by the Preference option are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters take precedence The default values of the parameters in the Tel Profile Settings and IP Profile Settings pages are identical to their default values in their respective primary configuration page If you modify a parameter in its primary configuration page or ini file that also appears in the profile pages the parameter s new value is automatically updated in the profile pages However once you modify any parameter in the profile pages modifications to parameters in the primary configuration pages or ini file no longer impact that profile pages 3 3 4 5 1 Configuring Coders The Coders page allows you to configure up to ten coders and their attributes for the device The first coder in the list has the highest priority and is used by the device whenever possible If the far end device cannot use the first coder the device attempts to use the next coder in the list and so on For a list of supported coders and for configuring coders using the ini file refer to the ini file parameter table CodersGroup described in SIP Confi
68. 3 Table 6 1 Table 6 2 hernet Pandas M ARVA a ae pe IP Neinor eee and VLAN Parameters Table 6 6 NF S Panen nde Table 6 7 DNS Parameters Table 6 8 DHCP Parameters i Table 6 9 NTP and Daylight Saving Time Parameters Table 6 10 General Web and Telnet Parameters Table 6 11 Web Parameters Table 6 12 Telnet Parameters Table 6 13 General Debugging and Diagnostic Parameters Table 6 14 Syslog CDR and Poe Parameters Table 6 15 RAI Parameters re Table 6 16 Serial Parameters Table 6 17 BootP Parameters Table 6 18 Table 6 19 Table 6 20 SIP User s Manual 12 Document LTRT 65413 Table je 6 Table 6 ne Dete Table Tabl Tabl Tabl Tabl Tabl Tabl Table Tabl Tabl Tabli Table Table B 1g Table 10 9 Multiple Interface Table Ex 7a u E tall AudioCodes MediaPack Series Table 10 107 Routing Table Example 1 d n duu ba o a ka aaa 461 Table 10 11 Multiple Interface Table Example 2 eee ee eee eee eee eee nene nt ent 462 Table 10 12 Routing Table Example Z ide l od innin nnana bob ce EEEE KE bl na 462 Table 10 13 Multiple Interface Table Example 3 eee eee eee eee nenene nen ent 463 Table 10 14 Routing Table Example 5 zi uiudiii nk bidnidklndvdk dudv k nnne nenie nan an e bad cista 463 Table 11 1 OMP Galactica celal crnce ibaa
69. 323 SIP call identifier Setup time in NTP format 1 The call s originator Answering IP or Originator PSTN Protocol type or family used on this leg of the call Connect time in NTP format 436 Value Format String up to 15 digits long Numeric Numeric Up to 32 octets Numeric Up to 32 octets String String String String Example Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc 5421385747 192 168 14 43 1 login Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Answer Originate etc VoIP Stop Acc Document LTRT 65413 SIP User s Manual Attribute Attribute Number Name H323 26 Disconnect Time H323 26 Disconnect Cause 26 H323 Gw ID 26 SIP Call ID Call 26 Terminator Called 9 Station ID Version 6 0 VSA No 29 30 33 34 35 Purpose Disconnect time in NTP format Q 931 disconnect cause code Name of the gateway SIP Call ID The call s terminator PSTN terminated call Yes IP terminated call No Destination phone number Calling Party Number ANI Account Request Type start or stop Note start isn t supported on the Calling Card application No of seconds tried in sending a particular record Number of octets received for that call duration Number of octets sent for that call duration
70. 38 Relay or 3 Fax Fallback 278 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 7 5 DTMF and Hook Flash Parameters The DTMF and hook flash parameters parameters are described in the table below Table 6 31 DTMF and Hook Flash Parameters Parameter Hook Flash Parameters Web EMS Hook Flash Code HookFlashCode Web EMS Hook Flash Option HookFlashOption Web Min Flash Hook Detection Period msec EMS Min Flash Hook Time MinFlashHookTime Version 6 0 Description Defines the digit pattern that when received from the Tel side indicates a Hook Flash event The valid range is a 25 character string The default is a null string Determines the hook flash transport type i e method by which hook flash is sent and received 0 Not Supported Hook Flash indication isn t sent default 1 INFO Sends proprietary INFO message with Hook Flash indication 4 RFC 2833 5 INFO Lucent Sends proprietary SIP INFO message with Hook Flash indication 6 INFO NetCentrex Sends proprietary SIP INFO message with Hook Flash indication The device sends the INFO message as follows Content Type application dtmf relay Signal 16 Where 16 is the DTMF code for hook flash 7 INFO HUAWAEI Sends a SIP INFO message with Hook Flash indication The device sends the INFO message as follows Content Length 17 Content Type application sscc event flas
71. 65535 The default is 47000 This ini file table parameter defines up to 16 NFS file systems so that the device can access a remote server s shared files and directories for loading cmp ini and auxiliary files using the Automatic Update mechanism As a file system the NFS is independent of machine types OSs and network architectures Note that an NFS file server can share multiple file systems There must be a separate row for each remote file system shared by the NFS file server that needs to be accessed by the device The format of this ini file table parameter is as follows NFSServers FORMAT NFSServers Index NFSServers_HostOrlP NFSServers_RootPath NFSServers_NfsVersion NFSServers_AuthType NFSServers_UID NFSServers_GID NFSServers VlanType NFSServers For example NFSServers 1 101 1 13 audio1 3 1 0 1 1 Notes You can configure up to 16 NFS file systems where the first index is 0 To avoid terminating current calls a row must not be deleted or modified while the device is currently accessing files on the remote NFS file system The combination of host IP and Root Path must be unique for each index in the table For example the table must include only one index entry with a Host IP of 192 168 1 1 and Root Path of audio This parameter is applicable only if VLANs are enabled or Multiple IPs is configured Fora detailed description of the table s parameters and to configure NFS
72. 9 1 Prefix to Add Field with Notation Stripped Digits Fram Index Destination Prefix Source Prefix Source IP Address Stripped Digits From Left Right Prefix to Add 1 549202000888 x 7 fos a5 In this configuration the following manipulation process occurs 1 the prefix is calculated 020215 in the example 2 the first seven digits from the left are removed from the original number in the example the number is changed to 8888888 3 the prefix that was previously calculated is then added SIP User s Manual 378 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 1 2 Digit Mapping The device collects digits until a match is found in the user defined digit pattern e g for closed numbering schemes or until a timer expires e g for open numbering schemes If a match is found or the timer expires the digit collection process is terminated The maximum number up to 49 of collected destination number digits that can be received i e dialed from the Tel side by the device can be defined using the parameter MaxDigits When the number of collected digits reaches the maximum or a digit map pattern is matched the device uses these digits for the called destination number Dialing ends and the device starts sending the digits when any of the following scenarios occur m Maximum number of digits is received m Inter digit timeout expires up to 10 seconds This is defined by using the
73. 9 28 Configuring Username and Password for Channels 5 8 in Authentication Page Gateway Port User Name Password Port 1 FXS Port 2 FXS Port 3 FXS Port 4 FX5 Port 5 FXS Port 6 FXS Port 7 FXS Port 8 FX5 7 Inthe Account Table page configure a single Account for Hunt Group ID 1 including an authentication user name and password and enable registration for this Account to ITSP 1 i e Serving IP Group is 1 Figure 9 29 Configuring Account for Registration to ITSP 1 Index ServedTrunkGroup ServingIPGroup Username Password HostName Register 1 rTsPtuser 1234 rse1 h SIP User s Manual 430 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 8 In the IP to Hunt Group Routing Table page configure that INVITEs with ITSP1 as the hostname in the From URI are routed to Hunt Group 1 and INVITEs with ITSP2 as the hostname in the From URI are routed to Hunt Group 2 In addition configure calls received from ITSP1 as associated with IP Group 1 Figure 9 30 Configuring ITSP to Hunt Group Routing IP Profile Source IPGroup 1D Hunt Dest Host Prefix Source Host Prefix Dest Phone Profi Source Phone Prefix Source IP Address Group A oa 10 ITSP1 r z 1 1 ITSP 9 In the Tel to IP Routing page configure Tel to IP routi
74. A e AudioCodes MediaPack Series 3 1 9 gt Toclose the Scenario mode 1 Simply click any tab besides the Scenarios tab on the Navigation bar or click the Cancel Scenarios button located at the bottom of the Navigation tree a message box appears requesting you to confirm exiting Scenario mode as shown below Figure 3 21 Confirmation Message Box for Exiting Scenario Mode Microsoft Internet Explorer 2 J This operation will cancel scenario mode are you sure 2 Click OK to exit Creating a Login Welcome Message You can create a Welcome message box alet message that appears after each successful login to the device s Web interface The ini file table parameter WelcomeMessage allows you to create the Welcome message Up to 20 lines of character strings can be defined for the message If this parameter is not configured no Welcome message box is displayed after login An example of a Welcome message is shown in the figure below Figure 3 22 User Defined Web Welcome Message after Login Microsoft Internet Explorer Gata t t t t i t a Welcome to the Embedded Web Server paata ta ob oe oe oko ke ok o oko a oko ok o k k ok kok ok eo kok o SACRA RAE ad RE RR a do ba EE ba o ba do EE da o EE d EE ba o d EE EE EE EEE EE EE EE EE HE o a SARA RR ERE EEE EERE ER EE REE REE EE EE ER EE ooo AE Table 3 2 ini File Parameter for Welcome Login Message Parameter Description WelcomeMessage Defines the Welcome mes
75. A UA requests the immediate removal of a binding by specifying an expiration interval of 0 for that contact address in a REGISTER reguest UA s should support this mechanism so 273 March 2010 Aa L tal AudioCodes MediaPack Series Parameter Description that bindings can be removed before their expiration interval has passed Use of the Contact header field value allows a registering UA to remove all bindings associated with an address of record AOR without knowing their precise values Note The REGISTER specific Contact header field value of applies to all registrations but it can only be used if the Expires header field is present with a value of 0 6 7 3 Voice Mail Parameters The voice mail parameters are described in the table below For detailed information on the Voice Mail application refer to the CPE Configuration Guide for Voice Mail Note Voice Mail is applicable only to FXO interfaces Table 6 29 Voice Mail Parameters Parameter Description Web EMS Voice Mail Interface Enables the device s Voice Mail application and VoiceMaillnterface determines the communication method used between the PBX and the device 0 None default 1 DTMF 2 SMDI Note To enable voice mail per Hunt Group you can use a Tel Profile ID that is configured with voice mail interface enabled This eliminates the phenomenon of call delay on lines not implementing voice mail when voice mail is enabled usin
76. AudioCodes IPMedia conferencing server Default 1 Non AudioCodes Media Server The Conference initiating INVITE sent by the device uses only the ConferencelD as the Request URI The conference server sets the Contact header of the 200 OK response to the actual unique identifier Conference URI to be used by the participants This Conference URI is then included by the device in the Refer To header value in the REFER messages sent by the device to the remote parties The remote parties join the conference by sending INVITE messages to the conference using this conference URI 2 On Board On board 3 way conference The conference is established on the device without the need for an external Conference server The device utilizes resources from idle ports to establish the conference call You can limit the number of simultaneous on board 3 way conference calls by using the parameter MaxInBoardConferenceCalls In addition you can designate ports that can t be used as a resource for on board conference calls initiated by other ports using the parameter 3WayConfNoneAllocateablePorts Version 6 0 303 March 2010 ca AudioCodes Parameter Web Max 3 Way Conference On Board Calls EMS Max In Board Calls MaxInBoardConferenceCalls Web Three Way Conference Non Allocatable Ports EMS Non Allocateable Port Number 3WayConfNoneAllocateablePorts Web Establish Conference Code EMS Establish Code ConferenceCode
77. Coders and Profile Definitions re 101 3 3 4 6 SIP Advanced Parameters s es Zd T Manipulation Tables sicaciccasccccidcaceainitabasinasstacsedsaweninadducertosavativans na 3 3 4 8 Routing Tables 3 3 4 9 Endpoint Settings 3 3 4 10 Configuring Endpoint Phone Numbers 3 3 4 11 SAS Parameters 3 3 5 Advanced Applications 3 3 5 1 Configuring Voice Mail Parameters 3 3 5 2 Configuring RADIUS Parameter 3 3 5 3 Configuring FXO Parameters 3 4 Management Tab PS 3 4 1 Management tonigi 3 4 1 1 Configuring the Management Settings 3 4 1 2 Configuring the oon oe 3 4 1 3 Maintenance Actions DE 3 4 2 Software Update EE EAEE E Mon ap 3 4 2 1 Loading Auxiliary Files E T 3 4 2 2 Loading a Software Upgrade Key 3 4 2 3 Software Upgrade Wizard E ee ouabain 3 4 2 4 Backing Up and Restoring Configuration a Chine N WE M 3 5 Status 8 Diagnostics Tab dm T 3 5 1 Status A Biagioni 3 5 1 1 Viewing the Device s Syslog Messages 172 3 5 1 2 Viewing Ethernet Port Information 174 3 5 1 3 Viewing Active IP Interfaces 174 3 5 1 4 Viewing Device Information 3 5 1 5 Viewing Performance Statistics P 35 1 6 Viewing Active AlAINS c 403xe06skdoacikaands aai IEF 3 5 2 Gateway Statistics is 3 5 2 1 Viewing Call Cour ters 3 5 2 2 Viewing SAS Registered Users 3 5 2 3 Viewing Call Routing Status RIOTO AE AE RAAE ME
78. Control traffic Gold Service class used for streaming applications Bronze Service class used for OAMP applications The Layer 2 Quality of Service parameters enables setting the values for the 3 priority bits in the VLAN tag of frames related to a specific service class according to the IEEE 802 1p standard The Layer 3 Quality of Service QoS parameters enables setting the values of the DiffServ field in the IP Header of the frames related to a specific service class The following QoS parameters can be set Table 10 5 Quality of Service Parameters Parameter Description Layer 2 Class Of Service Parameter VLAN Tag Priority Field VlanNetworkServiceClassPriority Sets the priority for the Network service class content Sets the priority for the Premium service class content VLANPremiumServiceClassMediaPriority media traffic Sets the priority for the Premium service class content VLANPremiumServiceClassControlPriority control traffic Sets the priority for the Gold service class content VLANGoldServiceClassPriority streaming traffic VLANBronzeServiceClassPriority Sets the priority for the Bronze service class content OAMP traffic Layer 3 Class Of Service Parameters TOS DiffServ NetworkServiceClassDiffServ Sets the DiffServ for the Network service class content Sets the DiffServ for the Premium service class content PremiumServiceClassMediaDiffServ i media traffic Sets the DiffServ for the Premium
79. DNS Table page item Figure 3 81 Internal DNS Table Page Domain Name First IP Address Second IP Address Third IP Address Fourth IP Address DomainName com 10 9 215 10 8 4 20 l 10 8 6 17 i 10 8 6 168 2 In the Domain Name field enter the host name to be translated You can enter a string of up to 31 characters long 3 In the First IP Address field enter the first IP address in dotted decimal format notation to which the host name is translated 4 Optionally in the Second IP Address Third IP Address and Second IP Address fields enter the next IP addresses to which the host name is translated 5 Click the Submit button to save your changes 6 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 8 6 Configuring the Internal SRV Table The Internal SRV Table page provides a table for resolving host names to DNS A Records Three different A Records can be assigned to each host name Each A Record contains the host name priority weight and port If the Internal SRV table is configured the device initially attempts to resolve a domain name using this table If the domain name isn t found the device performs an Service Record SRV resolution using an external DNS server You can also configure the Internal SRV table using the ini file table parameter SRV2IP refer to DNS Parameters on page 218 SIP User s Manual 134 Document LTRT 65413 SIP User
80. Endpoint Phone Number Table Page Phone Number Hunt Group ID Tel Profile ID 200 2 Configure the endpoint phone numbers according to the table below You must enter a number in the Phone Number fields for each port that you want to use 3 Click the Submit button to save your changes or click the Register or Un Register buttons to save your changes and to register unregister to a Proxy Registrar 4 To save the changes to the flash memory refer to Saving Configuration on page 161 Table 3 24 Endpoint Phone Number Table Parameters Parameter Description Channel s The device s channels or ports as labeled on the device s rear panel To enable channels enter the channel port numbers You can enter a range of channels by using the format n m where n represents the lower channel number and m the higher channel number e g 1 3 specifies channels ports 1 through 3 Version 6 0 143 March 2010 A K tal AudioCodes MediaPack Series Parameter Description Phone Number The telephone number that is assigned to the channel For a range of channels enter only the first telephone number Subsequent channels are assigned the next consecutive telephone number For example if you enter 400 for channels 1 to 4 then channel 1 is assigned phone number 400 channel 2 is assigned phone number 401 and so on These numbers are also used
81. Endpoint Phone Numbers on page 143 assign the phone numbers 101 to 104 to the device s endpoints Figure 9 20 Assigning Phone Numbers to Device 10 2 37 10 Channel s Phone Number Hunt Group ID 1 1 4 nor I 2 For the second device 10 2 37 20 in the Endpoint Phone Number Table page assign the phone numbers 201 to 204 to the device s endpoints Figure 9 21 Assigning Phone Numbers to Device 10 2 37 20 Channel s Phone Number Hunt Group ID 3 201 Configure the following settings for both devices In the Tel to IP Routing page refer to Configuring the Tel to IP Routing on page 126 add the following routing rules a In the first row enter 10 for the destination phone prefix and enter 10 2 37 10 for the destination IP address i e IP address of the first device b In the second row enter 20 for the destination phone prefix and 10 2 37 20 for the destination IP address i e IP address of the second device These settings enable the routing from both devices of outgoing Tel to IP calls that start with 10 to the first device and calls that start with 20 to the second device Figure 9 22 Routing Calls Between Devices Dest Phone Prefix Source Phone Prefix Dest IP Address gt Version 6 0 1023710 Bo E 2 20 10 2 37 20 Make a call Pick up the phone connected to port 1 of the first device and dial 102 to the p
82. Factor Basic RTP Packet Interval RFC 2833 TX Payload Type RFC 2633 RX Payload Type RFC 2198 Payload Type Fax Bypass Payload Type Enable RFC 3389 CN Payload Type SIP User s Manual 32 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To save configuration changes on a page to the device s volatile memory RAM m Click the Submit button which is located near the bottom of the page in which you are working modifications to parameters with on the fly capabilities are immediately applied to the device and take effect other parameters displayed on the page with the lightning symbol are not changeable on the fly and require a device reset refer to Resetting the Device on page 159 before taking effect Parameters saved to the volatile memory by clicking Submit revert to their previous settings after a hardware or software reset or if the device is powered down Therefore to ensure parameter changes whether on the fly or not are retained you need to save burn them to the device s non volatile memory i e flash refer to Saving Configuration on page 161 If you modify a parameter value and then attempt to navigate away from the page without clicking Submit a message box appears notifying you of this Click Yes to save your modifications or No to ignore them If you enter an invalid parameter value e g not in the range of permitted values and then click Submit a message
83. FaxModemBypassM The packing factor determines the number of coder payloads each the size of FaxModemBypassBasicRTPPacketinterval that are used to generate a single fax modem bypass packet When fax modem transmission ends the reverse switching from bypass coder to regular voice coder is performed To configure fax modem bypass mode perform the following configurations IsFaxUsed 0 FaxTransportMode 2 V21ModemTransportType 2 V22ModemTransportType 2 V23ModemTransportType 2 V32ModemTransportType 2 V34ModemTransportType 2 BellModemTransportType 2 Additional configuration parameters e FaxModemBypassCoderType e FaxBypassPayloadType e ModemBypassPayloadType Version 6 0 403 March 2010 c tal AudioCodes MediaPack Series 9 6 2 5 e FaxModemBypassBasicRTPPacketinterval e FaxModemBypassDJBufMinDelay Note When the device is configured for modem bypass and T 38 fax V 21 low speed modems are not supported and fail as a result When the remote non AudioCodes gateway uses G711 coder for voice and doesn t change the coder payload type for fax or modem transmission it is recommended to use the Bypass mode with the following configuration EnableFaxModemInbandNetworkDetection 1 FaxModemBypassCoderType same coder used for voice FaxModemBypassM same interval as voice ModemBypassPayloadType 8 if voice coder is A Law 0 if voice coder is Mu Law Fax Modem NSE Mode In this mode fax a
84. FaxRelayMaxRate G 711 Fax Modem Transport Mode In this mode when the terminating device detects fax or modem signals CED or AnsAM it sends a Re INVITE message to the originating device requesting it to re open the channel in G 711 VBD with the following adaptations m Echo Canceller off m Silence Compression off m Echo Canceller Non Linear Processor Mode off m Dynamic Jitter Buffer Minimum Delay 40 m Dynamic Jitter Buffer Optimization Factor 13 After a few seconds upon detection of fax V 21 preamble or super G3 fax signals the device sends a second Re INVITE enabling the echo canceller the echo canceller is disabled only on modem transmission A gpmd attribute is added to the SDP according to the following format m For G 711A law a gpmd 0 vbd yes ecan on or off for modems m For G 711 u law a gpmd 8 vbd yes ecan on or off for modems The parameters FaxTransportMode and VxxModemTransportType are ignored and automatically set to the mode called transparent with events To configure fax modem transparent mode set IsFaxUsed to 2 Fax Fallback In this mode when the terminating device detects a fax signal it sends a Re INVITE message to the originating device with T 38 If the remote device doesn t support T 38 replies with SIP response 415 Media Not Supported the device sends a new Re INVITE with G 711 VBD with the following adaptations m Echo Canceller on m Silence Compression
85. Gateway Configuration The procedure below describes how to configure the FXS interface at the remote PBX extension gt To configure the FXS interface 1 In the Endpoint Phone Numbers page refer to Configuring the Endpoint Phone Numbers on page 143 assign the phone numbers 100 to 107 to the device s endpoints Figure 9 11 Assigning Phone Numbers to FXS Endpoints Channel s Phone Number Hunt Group ID 1 8 100f 2 In the Automatic Dialing page refer to Automatic Dialing on page 137 enter the phone numbers of the FXO device in the Destination Phone Number fields When a phone connected to Port 1 off hooks the FXS device automatically dials the number 200 Figure 9 12 Automatic Dialing for FXS Ports Gateway Destination Phone Auto Dial Port Number Status FXS Enable FXS Enable FXS Enable FXS Enable FXS Enable FXS Enable v FXS Enable Vv FXS Enable W 3 In the Tel to IP Routing page refer to Configuring the Tel to IP Routing on page 126 enter 20 for the destination phone prefix and 10 1 10 2 for the IP address of the FXO device Figure 9 13 FXS Tel to IP Routing Configuration Dest Phone Prefix Source Phone Prefix R Dest IP Address 10 1 10 2 Note For the
86. Hop Count eguals 0 are local routes set automatically by the device Specifies the interface network type to which the routing rule is applied 0 OAMP default 1 Media 2 Control For detailed information on the network types refer to Configuring the Multiple Interface Table on page 50 Note For this parameter to take effect a device reset is required 6 1 4 Quality of Service Parameters The Quality of Service QoS parameters are described in the table below The device allows you to specify values for Layer 2 and Layer 3 priorities by assigning values to the following service classes m Network Service class network control traffic ICMP ARP Premium Media service class used for RTP Media traffic Premium Control Service class used for Call Control traffic Gold Service class used for streaming applications Bronze Service class used for OAMP applications The Layer 2 QoS parameters enables setting the values for the 3 priority bits in the VLAN tag of frames related to a specific service class according to the IEEE 802 1p standard The Layer 3 QoS parameters enables setting the values of the DiffServ field in the IP Header of the frames related to a specific service class SIP User s Manual 212 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Table 6 4 QoS Parameters Parameter Description Layer 2 Class Of Service Parameters VLAN
87. ID IP2IPRouting_DestIPGroupID Version 6 0 Determines the destination type to which the outgoing INVITE is sent 0 IP Group default The INVITE is sent to the IP Group s Proxy Set if the IP Group is of SERVER type registered contact from the database if USER type 1 DestAddress The INVITE is sent to the address configured in the following fields Destination Address Destination Port and Destination Transport Type 2 Request URI The INVITE is sent to the address indicated in the incoming Request URI If the fields Destination Por and Destination Transport Type are configured the incoming Request URI parameters are overridden and these fields take precedence 3 ENUM An ENUM query is sent to conclude the destination address If the fields Destination Port and Destination Transport Type are configured the incoming Request URI parameters are overridden and these fields take precedence The IP Group ID to where you want to route the call The INVITE messages are sent to the IP address es defined for the Proxy Set associated with this IP Group If you select an IP Group it is unnecessary to configure a destination IP address in the Destination Address field However if both parameters are configured the IP Group takes precedence If the destination IP Group is of USER type the device searches for a match between the Request URI of the received INVITE to an A
88. IP Groups on page 91 configure the two IP Groups 1 and 2 Assign Proxy Sets 1 and 2 to IP Groups 1 and 2 respectively Figure 9 25 Configuring IP Groups 1 and 2 in the IP Group Table Page v Common Parameters Type Description Proxy Set ID SIP Group Name Contact User Version 6 0 429 March 2010 7a T c tal AudioCodes MediaPack Series 4 In the Endpoint Phone Number Table page configure Hunt Group ID 1 for channels 1 4 and Hunt Group ID 2 for channels 5 8 Figure 9 26 Assigning Channels to Hunt Groups Channel s JE Phone Number Hunt Group ID Tel Profile ID 1 4 5 6 5 In the Hunt Group Settings page configure Per Account registration for Hunt Group ID 1 without serving IP Group and associate it with IP Group 1 Configure Per Endpoint registration for Hunt Group ID 2 and associated it with IP Group 2 Figure 9 27 Configuring Registration Mode for Hunt Groups and Assigning to IP Group Serving Hunt Channel Select Mode Registration Group ID Mode Gateway Name Contact User 1 Cyclic Ascending v Per Account Y 1 Mi 2 Cyclic Ascending v Per Endpoint v 2 v 6 In the Authentication page for channels 5 8 i e Hunt Group ID 2 define for each channel the registration authentication user name and password Figure
89. IP Profile IPProfile SIP User s Manual This ini file table parameter configures the IP Profile table Each IP Profile ID includes a set of parameters which are typically configured separately using their individual global parameters You can later assign these IP Profiles to Tel to IP routing rules Prefix parameter IP to Tel routing rules PSTNPrefix parameter and IP Groups IPGroup parameter The format of this parameter is as follows IPProfile 286 Document LTRT 65413 SIP User s Manual Parameter Version 6 0 6 Configuration Parameters Reference Description FORMAT IPProfile_Index IPProfile_ProfileName IPProfile IpPreference IPProfile CodersGroupID IPProfile_IsFaxUsed IPProfile JitterBufMinDelay IPProfile JitterBufOptFactor IPProfile IPDiffServ IPProfile SiglPDiffServ IpProfile SCE IPProfile RTPRedundancyDepth IPProfile RemoteBaseUDPPort IPProfile CNGmode IPProfile VxxTransportType IPProfile NSEMode IpProfile IsDTMFUsed IPProfile PlayRBTone2IP IPProfile EnableEarlyMedia IPProfile ProgressIndicator2IP IPProfile EnableEchoCanceller IPProfile CopyDest2RedirectNumber IPProfile MediaSecurityBehaviour IPProfile CallLimit IPProfile DisconnectOnBrokenConnection IPProfile FirstTxDtmfOption IPProfile SecondTxDtmfOption IPProfile RxDTMFOption IpProfile EnableHold IpProfile InputGain IpProfile VoiceVolume IpProfile AddlElnSetup IpProfile SBCExtensionCodersGroupID IPProfile Media
90. LeaveFromRight SourceNumberMapTel2Ip Prefix2Add SourceNumberMapTel2Ip Suffix2Add SourceNumberMapTel2Ip IsPresentationRestricted NumberMapTel2Ip SrcTrunkGroupID NumberMapTel2lp_SrclPGroupID SourceNumberMapTel2Ip For example SourceNumberMapTel2Ip 0 22 03 0 0 2 667 0 SourceNumberMapTel2Ip 0 10 10 255 255 3 0 5 100 255 Notes This table parameter can include up to 20 indices The parameters NumberType and NumberPlan are not applicable RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType NumberPlan and IsPresentationRestricted are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and then Suffix2Add An asterisk represents all IP addresses IsPresentationRestricted is set to Restricted only if Asserted Identity Mode is set to P Asserte To configure manipulation of source numbers for Tel to IP calls using the Web interface refer to Configuring the Number Manipulation Tables on page 115 Fora description on using ini file table parameters refer to to Configuring ini File Table Parameters on page 186 347 March 2010 A K e AudioCodes MediaPack Series Parameter Description Web Source Phone Number Manipulation Table for IP to Tel Cal
91. MP 118 Coders G723 G729 G728 NETCODER GSM FR GSM EFR AMR EVRC OCELP G727 ILBC EVRC B AMR WB G722 EG711 M5 RTA NB DSP Voice features Channel Type RTP ATM PCI DspCh 30 IPMediaDspCh 30 E1Trunks 84 TiTrunks 84 FX5Ports 24 FXOPorts 24 Control Protocols MGCP MEGACO H323 SIP Default features Coders G711 G726 Add a Software Upgrade Key Add Key Send Upgrade Key file from your computer to the device Browse J SenaFie Reset with flash burn is reguired after file is loaded 2 Backup your current Software Upgrade Key as a precaution so that you can re load this backup key to restore the device s original capabilities if the new key doesn t comply with your requirements a Inthe Current Key field copy the string of text and paste it in any standard text file b Save the text file to a folder on your PC with a name of your choosing 3 Open the new Software Upgrade Key file and ensure that the first line displays ILicenseKeys and that it contains one or more lines in the following format S N lt serial number gt lt long Software Upgrade Key gt For example S N370604 jCx6r5tovClIKaBBbhPtT53Yj One S N must match the serial number of your device The device s serial number can be viewed in the Device Information page refer to Viewing Device Information on page 174 4 Follow one of the following procedures depending on whether you are loading a single
92. MediaPack Series m Cadence A repeating sequence of on and off sounds Up to four different sets of on off periods can be specified m Burst A single sound followed by silence Only the First Signal On time and First Signal Off time should be specified All other on and off periods must be set to zero The burst tone is detected after the off time is completed You can specify several tones of the same type These additional tones are used only for tone detection Generation of a specific tone conforms to the first definition of the specific tone For example you can define an additional dial tone by appending the second dial tone s definition lines to the first tone definition in the ini file The device reports dial tone detection if either of the two tones is detected The Call Progress Tones section of the ini file comprises the following segments m NUMBER OF CALL PROGRESS TONES Contains the following key Number of Call Progress Tones defining the number of Call Progress Tones that are defined in the file m CALL PROGRESS TONE X containing the Xth tone definition starting from 0 and not exceeding the number of Call Progress Tones less 1 defined in the first section e g if 10 tones then it is 0 to 9 using the following keys SIP User s Manual Tone Type Call Progress Tone types 1 Dial Tone 2 Ringback Tone 3 Busy Tone 7 Reorder Tone 8 Confirmation Tone 9 Call Waiting Tone he
93. NO Configure the SNMP community strings parameters according to the table below Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 Note To delete a community string select the Delete check box corresponding to the community string that you want to delete and then click Submit 155 March 2010 7a tal AudioCodes MediaPack Series Table 3 27 SNMP Community Strings Parameters Description Parameter Description Community String Read Only SNMPReadOnlyCommunityString_x Up to five read only community strings up to 19 characters each The default string is public Read Write SNMPReadWriteCommunityString x Up to five read write community strings up to 19 characters each The default string is private Trap Community String Community string used in traps up to 19 characters SNMPTrapCommunityString The default string is trapuser 3 4 1 1 3 Configuring SNMP V3 Users The SNMP V3 Settings page allows you to configure authentication and privacy for up to 10 SNMP v3 users gt To configure the SNMP v3 users 1 Access the Management Settings page as described in Configuring the Management Settings on page 152 2 In the SNMP V3 Table field click the right pointing arrow button the SNMP V3 Settings page appears Figure 3 98 SNMP V3 Setting Page Add Apply User Name Authentication Protoc
94. Name Paranie 6 17 2 Automatic Update Parameters bs Restoring Factory Default a TA hostel zi k o E E AE slat ficial duha oa bead aan ja Aa dk EE o A Ti 73 Ractotina Defaults using q Haea Reset Butto PS nOD Auxiliary Configuration Files cccccsssssceeesssseeeeeessseeeeeeessseeeeeeesseeeeeeneaas 367 8 1 Call Progress Tones File ji sinks banimin T 8 1 1 Distinctive Ringing 370 8 1 2 FXS Distinctive Ringing and Call Waiting Tones per Source Number iinan lt 19 SIP User s Manual 6 Document LTRT 65413 SIP User s Manual Contents 8 2 8 3 8 4 9 1 Version 6 0 Prerecorded Tones File Dial Plan Fil mik User Information File Routing Applications 92 1 em ity and QoS 7 March 2010 7a 4 wl AudioCodes MediaPack Series 9 7 7 Message Waiting Indication nussii iain oasian aana eee eee aaia a s 416 9 7 8 Caller ID Bos ER rrr ter AT 9 7 8 1 Caller ID Detection Gener lon c on n the Tel OIG z a be a E DO 9 7 8 2 Debugging a Caller ID Detection on FXO 9 7 8 3 Caller ID on the IP Side Pe 9 7 9 Three Way Conferencing ee 9 8 Routing Examples eee 9 8 1 SIP Call Flow Poa dakou ko ana PR 9 8 2 SIP Authentication Example PM 9 8 3 Proxy or Registrar Registration Example 9 8 4 Establishing a Call between Two Devices ee 9 8 5 SIP Trunking between Enterprise and ITSPs AAA NTE AA PE OOP O PO 9 9 Mapping
95. Name for OPTIONS UseGatewayNameF orOptions Web EMS User Name UserName SIP User s Manual MediaPack Series Description Enables the use of DNS Naming Authority Pointer NAPTR and Service Record SRV queries to discover Proxy servers 0 A Record default 1 SRV 2 NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy IP address parameter contains a domain name without port definition e g ProxylP domain com an SRV guery is performed The SRV guery returns up to four Proxy host names and their weights The device then performs DNS A record gueries for each Proxy host name according to the received weights to locate up to four Proxy IP addresses Therefore if the first SRV guery returns two domain names and the A record gueries return two IP addresses each no additional searches are performed If set to NAPTR 2 an NAPTR query is performed If it is successful an SRV guery is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy IP address parameter contains a domain name with port definition e g ProxylP domain com 5080 the device performs a regular DNS A record guery If a specific Transport Type is defined a NAPTR query is not performed Note When enabled NAPTR SRV queries are used to discover Proxy servers even if
96. Note Although not recommended you can use both default Proxy Set ID 0 and IP Groups for call routing For example on the Hunt Group Settings page refer to Configuring Hunt Group Settings on page 85 you can configure a Serving IP Group to where you want to route specific Hunt Group s endpoints while all other device endpoints use the default Proxy Set At the same you can also use IP Groups in the Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 to configure the default Proxy Set if the parameter PreferRouteTable is setto 1 To summarize if the default Proxy Set is used the INVITE message is sent according to the following preferences To the Hunt Group s Serving IP Group ID as defined in the Hunt Group Settings table SIP User s Manual 98 Document LTRT 65413 SIP User s Manual 3 Web Based Management Parameter Description According to the Tel to IP Routing if the parameter PreferRouteTable is set to 1 To the default Proxy Typically when IP Groups are used there is no need to use the default Proxy and all routing and registration rules can be configured using IP Groups and the Account tables refer to Configuring the Account Table on page 93 Proxy Address The IP address and optionally port number of the Proxy server Proxylp IpAddress Up to five IP addresses can be configured per Proxy Set Enter the IP address as an FQDN or in dotted decimal notation e g
97. O0 Normal 1 Reversed 2 M A Line polarity 0 Positive 1 Megative Message Waiting Indication 0 Off 1 On o olololololol o 3 To view RTP RTCP or voice settings click the relevant button Version 6 0 49 March 2010 A e AudioCodes MediaPack Series 3 3 3 3 1 3 3 1 1 Configuration Tab The Configuration tab on the Navigation bar displays menus in the Navigation tree related to device configuration These menus include the following m Network Settings refer to Network Settings on page 50 m Media Settings refer to Media Settings on page 60 m Security Settings refer to Security Settings on page 66 m Protocol Configuration refer to Protocol Configuration on page 83 m Advanced Applications refer to Advanced Applications on page 148 Network Settings The Network Settings menu allows you to configure various networking parameters This menu includes the following items m P Settings refer to Configuring the Multiple Interface Table on page 50 m Application Settings refer to Configuring the Application Settings on page 54 m P Routing Table refer to Configuring the IP Routing Table on page 58 mM QOS Settings refer to Configuring the QoS Settings on page 60 Configuring the Multiple Interface Table The Multiple Interface Table page allows you to configure up to 16 logical network interfaces each with its own
98. PSTN Release Cause to SIP Rosne e T 9 10 Querying Device Channel Resources using SIP OPTIONS 9 11 Event Notification using X Detect Header i Seer 9 12 Supported RADIUS A Trice ao nn 9 13 Call Detail Record aries E PE E E E EAE ET EEE 9 14 RTP Multiplexing ThroughPacket 9 15 Duna ier Bitte Herein aaaeaii aA a AEEA 10 Networking Capabilities ccd cnededenconndenneonaneeuis 443 10 1 Ethernet Interface Configuration PEE 10 2 m Address Translation Support j j 2 2 Firet incum Pa ket Mechanism Satake du E E 5 10 2 3 No Op PACKEES T 10 3 IP Multicasting TAE ETE APE ce E PE E eee o o ed 10 4 Robust Raia of Media StreamS ecceeerrieeeerrrrrrresesrrrrrrrrnsnsrrrr 446 10 5 Multiple Routers Support i PEE T PET PEE i 10 6 Simple Network Time Pralea ASit P E ee K As 10 7 IP QoS via Differentiated Services DiffSONV 948 10 8 Network Configuration nr Sinaia d ALBA 10 8 1 Multiple Network interia s a VLANs debut ix il AP N AE E EET 10 8 1 1 Overview of Multiple Interface TAB 449 10 8 1 2 Columns of the Multiple Interface Table 450 10 8 1 3 Other Related Parameters za 452 10 8 1 4 Multiple Interface Table Configuration Summ ry and Guidelines 455 10 8 1 5 Troubleshooting the Multiple Interface Table 456 10 8 2 Routing Table gt 457 10 8 2 1 Routing Tabl Ove jew 457 10 8 2 2 Routing Table Columns 457 10 8 2 3 Routing Table Configuration 2 and Guidelines Mien 10 8 2 4 Troublesho
99. Port TrmSd TrmReason Fax InPackets OutPackets PackLoss RemotePackLoss Uniqueld SetupTime Version 6 0 Table 9 6 Supported CDR Fields Description Report for either Call Started Call Connected or Call Released Port Number SIP Call Identifier Physical Trunk Number always set to 1 as not applicable Selected B Channel always set to 0 as not applicable SIP Conference ID Trunk Group Number Endpoint Type Call Originator IP Tel Source IP Address Destination IP Address Source Phone Number Type Source Phone Number Plan Source Phone Number Source Number Before Manipulation Destination Phone Number Type Destination Phone Number Plan Destination Phone Number Destination Number Before Manipulation Call Duration Selected Coder Packet Interval RTP IP Address Remote RTP Port Initiator of Call Release IP Tel Unknown Termination Reason Fax Transaction during the Call Number of Incoming Packets Number of Outgoing Packets Local Packet Loss Number of Outgoing Lost Packets unique RTP ID Call Setup Time 439 March 2010 ca AudioCodes MediaPack Series Field Name Description ConnectTime Call Connect Time ReleaseTime Call Release Time RTPdelay RTP Delay RTPjitter RTP Jitter RTPssrc Local RTP SSRC RemoteRTPssrc Remote RTP SSRC RedirectReason Redirect Reason TON Redirection Phone Number Type MeteringPulses Number of Generated Metering Pulses NPI Redirection Phone Number Plan Redir
100. Port 6 provides lifeline to FXS Port 2 and so on Upon power outage and or network failure PSTN connectivity is maintained for the FXS phone user 0 Lifeline is activated upon power failure default 1 Lifeline is activated upon power failure or when the link is down physically disconnected 2 Lifeline is activated upon power failure when the link is down or upon network failure logical link disconnected Notes For this parameter to take effect a device reset is required This parameter is applicable only to FXS interfaces To enable Lifeline switching on network failure the LAN watch dog must be activated i e set the parameter EnableLANWatchDog to 1 For a detailed description on cabling the device for Lifeline refer to the device s Installation Manual Defines the time interval in seconds that the device s operation is delayed after a reset The valid range is 0 to 45 The default value is 7 seconds Note This feature helps overcome connection problems caused by some LAN routers or IP configuration parameters modifications by a DHCP server 6 3 2 Syslog CDR and Debug Parameters The Syslog CDR and debug parameters are described in the table below Table 6 14 Syslog CDR and Debug Parameters Parameter Web EMS Syslog Server IP Address SyslogServerlP Web Syslog Server Port EMS Syslog Server Port Number SyslogServerPort Web Enable Syslog EMS Syslog enable
101. Realm KI KI a Coders Group 1 TLS Re Handshake Interval 0 W K K a Coders Group 2 a Coders Group 3 a TLS Remote Subject Name a Coders Group 4 rl PeerHostName Verification Mode Disable v 4 IP Profile Verify Server Certificate KI a g W gl K a Telephony Profile a SRTP Offered Suites Sa 3 From the SRTP Offered Suites SRTPofferedSuites drop down list select one of the crypto suites Version 6 0 199 March 2010 Aa e AudioCodes MediaPack Series 5 7 Provisioning SIP MLPP Parameters This section describes how to configure the MLPP Multi Level Precedence and Preemption parameters using the EMS gt To configure the MLPP parameters 1 In the MG Tree select the device that you want to configure a graphical representation of the device is displayed in the main pane 2 Open the MLPP screen Configuration icon gt SIP Advanced Configuration menu gt MLPP tab Parameters List a General Features 1 Call Priority Mode Disable g a General Features 2 amp Kl a Transport Info 2m Default Name Space DSN a Tones And Progress M E Default Call Priority 0 Diff Serv 50 a Voice Mail K B a Emergency Preemption Tone Duration 3 a Debug a MLPP T a Stand Alone Survivability a Conference a Default Service Domain ooooo0 g Normalized Serice Domain 000000 a RTP DSCP for MLPP R
102. Redirect Number CopyDest2RedirectNumber Version 6 0 Description Determines whether the device copies the called number to the outgoing SIP Diversion header for Tel to IP calls Therefore the called number is used as a redirect number Call redirection information is typically used for Unified Messaging and voice mail services to identify the recipient of a message 0 Don t copy Disable default 1 Copy after phone number manipulation Copies the called number after manipulation The device first performs Tel to IP destination phone number manipulation i e on the SIP To header and only then copies the manipulated called number to the SIP Diversion header for the Tel to IP call Therefore with this option the called and redirected numbers are identical 2 Copy before phone number manipulation Copies the called number before manipulation The device first copies the original called number to the SIP Diversion header and then performs Tel to IP destination phone number manipulation Therefore this allows you to have different numbers for the called i e SIP To header and redirected i e SIP Diversion header numbers Notes This parameter can also be configured for IP Profiles using the parameter IPProfile 341 March 2010 ca AudioCodes Parameter Web Redirect Number Tel gt IP EMS Redirect Number Map Tel to IP RedirectNumberMapTel2IP Phone Context Parameters Web EMS Add
103. Syslog messages may increase the network traffic To configure Syslog logging levels use the parameter GwDebugLevel For information on the Syslog refer to the Product Reference Manual Logs are also sent to the RS 232 serial port For information on establishing a serial communications link with the device refer to the device s Installation Manual Determines the method used for Syslog messages 0 Send all Syslog messages to the defined Syslog server default 1 Send all Syslog messages using the Debug Recording mechanism 2 Send only Error and Warning level Syslog messages using the Debug Recording mechanism For a detailed description on Debug Recording refer to the Product Reference Manual The maximum size in bytes threshold of logged Syslog messages bundled into a single UDP packet after which they are sent toa Syslog server The valid value range is 0 to 1220 where 0 indicates that no bundling occurs The default is 1220 Note This parameter is applicable only if the GWDebugLevel parameter is set to 7 Defines the destination IP address to where CDR logs are sent The default value is a null string which causes CDR messages to be sent with all Syslog messages to the Syslog server Notes The CDR messages are sent to UDP port 514 default Syslog port This mechanism is active only when Syslog is enabled i e the parameter EnableSyslog is set to 1 Determines whether Ca
104. Table TxDTMFOption DisableAutoDTMFMute Version 6 0 This ini file table parameter configures up to two preferred transmit DTMF negotiation methods The format of this parameter is as follows TxDTMF Option FORMAT TxDTMFOption_Index TxDTMFOption Type TxDTMF Option For example TxDTMFOption 0 1 TxDTMFOption 1 3 Notes This parameter can include up two indices For a description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Enables disables the automatic muting of DTMF digits when out of band DTMF transmission is used 0 Automatic mute is used default 1 No automatic mute of in band DTMF When this parameter is set to 1 the DTMF transport type is set according to the parameter DTMFTransportType and the DTMF digits aren t muted if out of band DTMF mode is selected TxDTMFOption set to 1 2 or 3 This enables the sending of DTMF digits in band transparent of RFC 2833 in addition to out of band DTMF messages Note Usually this mode is not recommended 281 March 2010 ca AudioCodes Parameter Web EMS Enable Digit Delivery to IP EnableDigitDelivery2IP Web Enable Digit Delivery to Tel EMS Enable Digit Delivery EnableDigitDelivery Web EMS RFC 2833 Payload Type RFC2833PayloadType SIP User s Manual MediaPack Series Description The Digit Delivery feature enables sending DTMF digits to the destinati
105. Table on page 97 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Enables the device to register to a Proxy Registrar server 0 Disable The device doesn t register to Proxy Registrar server default 1 Enable The device registers to Proxy Registrar server when the device is powered up and at every user defined interval configured by the parameter RegistrationTime Note The device sends a REGISTER reguest for each channel or for the entire device according to the AuthenticationMode parameter Registrar domain name If specified the name is used as the Reguest URI in REGISTER messages If it isn t specified default the Registrar IP address or Proxy name or IP address is used instead The valid range is up to 49 characters The IP address or FQDN and port number optional of the Registrar server The IP address is in dotted decimal notation e g 201 10 8 1 lt 5080 gt Notes If not specified the REGISTER request is sent to the primary Proxy server When a port number is specified DNS NAPTR SRV queries aren t performed even if the parameter DNSQueryType is set to 1 or 2 If the parameter RegistrarlP is set to an FQDN and is resolved to multiple addresses the device also provides real time switching hotswap mode between different Registrar IP addresses the parameter IsProxyHotSwap is set to 1 If the first Registr
106. Tag Priority Field Web Network Priority EMS Network Service Class Priority VLANNetworkServiceClassPriority Web Media Premium EMS Premium Service Class Media Priority Priority VLANPremiumServiceClassMediaPriority Web Control Premium Priority EMS Premium Service Class Control Priority VLANPremiumServiceClassControlPriority Web Gold Priority EMS Gold Service Class Priority VlanGoldServiceClassPriority Web Bronze Priority EMS Bronze Service Class Priority VLANBronzeServiceClassPriority Defines the VLAN priority IEEE 802 1p for Network Class of Service CoS content The valid range is 0 to 7 The default value is 7 Defines the VLAN priority IEEE 802 1p for the Premium CoS content and media traffic The valid range is 0 to 7 The default value is 6 Defines the VLAN priority IEEE 802 1p for the Premium CoS content and control traffic The valid range is 0 to 7 The default value is 6 Defines the VLAN priority IEEE 802 1p for the Gold CoS content The valid range is 0 to 7 The default value is 4 Defines the VLAN priority IEEE 802 1p for the Bronze CoS content The valid range is 0 to 7 The default value is 2 Layer 3 Class of Service TOS DiffServ Parameters For detailed information on IP QoS via Differentiated Services refer to IP QoS via Differentiated Services DiffServ on page 448 Web Network QoS EMS Network Service Class Diff Serv NetworkServiceClassDiffServ Web M
107. The Alternative Routing feature is disabled but read only information on the QoS of the destination IP addresses is provided For information on the Alternative Routing feature refer to Configuring Alternative Routing Based on Connectivity and QoS on page 399 337 March 2010 A tal AudioCodes MediaPack Series Parameter Description Web Alt Routing Tel to IP Mode Determines the event s reason for triggering Alternative EMS Alternative Routing Mode Routing AltRoutingTel2IPMode 0 None Alternative routing is not used 1 Connectivity Alternative routing is performed if a ping to the initial destination fails 2 QoS Alternative routing is performed if poor QoS is detected 3 Both Alternative routing is performed if either ping to initial destination fails poor QoS is detected or the DNS host name is not resolved default Notes QoS is quantified according to delay and packet loss calculated according to previous calls QoS statistics are reset if no new data is received within two minutes For information on the Alternative Routing feature refer to Configuring Alternative Routing Based on Connectivity and QoS on page 399 To receive quality information displayed in the Quality Status and Quality Info fields in Viewing IP Connectivity on page 183 per destination this parameter must be set to 2 or 3 Web Alt Routing Tel to IP Determines the method used by the device
108. The default gateway s address must be on the same subnet as the interface address In addition the default gateway can only be configured on one of the interfaces running Media traffic A separate routing table allows configuring additional routing rules Refer to Routing Table on page 457 for more details The default gateway configured in the example below 200 200 85 1 is available for the applications allowed on that interface Media 8 Control Outgoing management traffic originating on interface 0 is never directed to this default gateway Table 10 3 Configured Default Gateway Example Application Interface Prefix VLAN Interface Index Type Mode IP Address Length Gateway ID Name 0 OAMP IPv4 192 168 85 14 16 0 0 0 0 100 Mgmt Manual P U U g 1 Media IPv4 200 200 85 14 24 200 200 85 1 200 CntriMedia Control Manual Version 6 0 451 March 2010 e AudioCodes MediaPack Series A separate routing table allows configuring routing rules Configuring the following routing rule enables OAMP applications to access peers on subnet 17 17 0 0 through the gateway 192 168 0 1 Table 10 4 Separate Routing Table Example Destination Prefix Length Subnet Mask Gateway Interface Metric 17 17 0 0 16 192 168 0 1 0 1 10 8 1 2 6VLAN ID Column This column defines the VLAN ID for each interface When using VLANs this column must hold a unique value for each interface of the same address family 10 8 1 2 7
109. This is sometimes useful for example when the device FXO is connected to a PBX and the communication between the two can t be disconnected e g when using reverse polarity 48 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To reset a channel Click the required FXS or FXO port icon and then from the shortcut menu choose Reset Channel the channel is changed to inactive i e the port icon is displayed in grey Figure 3 30 Reset Channel Example MP 11x Uplink Ready P ower 3 2 3 Viewing Analog Port Information The Home page allows you to view detailed information on a specific FXS or FXO analog port such as RTP RTCP and voice settings gt To view detailed port information 1 Click the port for which you want to view port settings the shortcut menu appears Figure 3 31 Port Settings Example MP 11x Port Settings Uplink Ready Power 2 From the shortcut menu click Port Settings the Basic Channel Information screen appears Figure 3 32 Basic Channel Information Page SIP Basic RTP RTCP Voice Settings v Channel Identifier 4 Status Inactive Call ID 0 Endpoint ID Call Duration sec 0 Call Type Voice Call Destination 10 13 4 13 Coder G7114law_64 Last Current Disconnect Duration 0 Line Current m Line Voltage V Hook 0 Onhook 1 Off hook Ring 0 Off 1 On Line Connected 0 Disconnected 1 Connected Polarity state
110. Type the text rather than copy and paste Save the IKE pre shared key as later on you need to enter the same value in the EMS when defining the device For more information on CLI refer to the Product Reference Manual For more information on securing communication protocols refer to the EMS Users Manual gt To configure the device for communicating via IPSec with the EMS 1 Open an SSH Client session e g PuTTY and then connect to the device e Ifa message appears with the RSA host key click Yes to continue e The default username and password are Admin case sensitive Verify that the shell prompt appears 2 Type Conf and then press Enter CONFiguration gt 3 Type cf set and then press Enter the following prompt is displayed Enter data below Type a period on an empty line to finish The configuration session is now active and all data entered at the terminal is parsed as configuration text formatted as an ini file 4 Type the following at the configuration session IPsecSATable FORMAT IPsecSATable Index IPsecSATable RemoteEndpointAddressOrName IPsecSATable AuthenticationMethod IPsecSATable SharedKey IPsecSATable SourcePort IPsecSATable DestPort IPsecSATable Protocol IPsecSATable PhaselSaLifetimeInSec IPsecSATable Phase2SaLifetimeInSec IPsecSATable Phase2SaLifetimeInKB IPsecSATable DPDmode IPsecSATable IPsecMode IPsecSATable RemoteTunnelAddress IPsec
111. User Agent for subsequent requests SIP User s Manual 180 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 5 2 3 Viewing Call Routing Status The Call Routing Status page provides you with information on the current routing method used by the device This information includes the IP address and FQDN if used of the Proxy server with which the device currently operates gt To view the call routing status m Open the Call Routing Status page Status amp Diagnostics tab gt Gateway Statistics menu gt Calls Routing Status page item Figure 3 116 Call Routing Status Page Call Routing Method Proxy GK w Active Proxy Sets Status E IP Address _ 10 13 4 6 10 13 4 6 m Table 3 35 Call Routing Status Parameters Parameter Description Call Routing Method Proxy GK Proxy server is used to route calls Routing Table The Tel to IP Routing is used to route calls IP Address Not Used Proxy server isn t defined IP address and FQDN if exists of the Proxy server with which the device currently operates State N A Proxy server isn t defined OK Communication with the Proxy server is in order Fail No response from any of the defined Proxies Version 6 0 181 March 2010 7a K tal AudioCodes MediaPack Series 3 5 2 4 Viewing Registration Status The Registration Status page displays whether t
112. VolP Device Hunt LA Group ID 2 POTS Phones ITSP 2 IP Group 2 Proxy Set 2 IP 10 8 8 40 IP 10 8 8 10 X PSTN Network SIP User s Manual 428 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities gt To configure call routing between an Enterprise and two ITSPs 1 Enable the device to register to a Proxy Registrar server using the parameter IsRegisterNeeded 2 In the Proxy Sets Table page refer to Configuring the Proxy Sets Table on page 97 configure two Proxy Sets and for each enable Proxy Keep Alive using SIP OPTIONS and round robin load balancing method e Proxy Set 1 includes two IP addresses of the first ITSP ITSP 1 10 33 37 77 and 10 33 37 79 and using UDP e Proxy Set 2 includes two IP addresses of the second ITSP ITSP 2 10 8 8 40 and 10 8 8 10 and using TCP The figure below displays the configuration of Proxy Set ID 1 Perform similar configuration for Proxy Set ID 2 but using different IP addresses Figure 9 24 Configuring Proxy Set ID 1 in the Proxy Sets Table Page v Proxy Set ID Proxy Address Transport Type 10 33 37 77 UDP Y 10 33 37 79 Iv Enable Proxy Keep Alive Using Options Proxy Keep Alive Time 60 Proxy Load Balancing Method Round Robin Is Proxy Hot Swap No 3 In the IP Group Table page refer to Configuring the
113. Web Echo Canceler EMS Echo Canceller Enable EnableEchoCanceller EMS Echo Canceller Hybrid Loss ECHybridLoss ECNLPMode EchoCancellerAggressiveNLP Web Enable RFC 3389 CN Payload Type EMS Comfort Noise Enable EnableStandardSIDPayloadType RTPSIDCoeffNum SIP User s Manual MediaPack Series Description 2 Enable without Adaptation A single silence packet is sent during a silence period applicable only to G 729 Note If the selected coder is G 729 the value of the annexb parameter of the fmtp attribute in the SDP is determined by the following rules f EnableSilenceCompression is 0 annexb no If EnableSilenceCompression is 1 annexb yes If EnableSilenceCompression is 2 and IsCiscoSCEMode is 0 annexb yes If EnableSilenceCompression is 2 and IsCiscoSCEMode is 1 annexb no Determines whether echo cancellation is enabled and therefore echo from voice calls is removed 0 Off Echo Canceler is disabled 1 On Echo Canceler is enabled default Note This parameter is used to maintain backward compatibility Sets the four wire to two wire worst case Hybrid loss the ratio between the signal level sent to the hybrid and the echo level returning from the hybrid 0 6 dB default 1 N A 2 0dB 3 3dB Defines the echo cancellation Non Linear Processing NLP mode 0 NLP adapts according to echo changes default 1 Disable
114. Web Enable Microsoft Extension EnableMicrosofExt SIP User s Manual MediaPack Series Description For Analog FXS FXO interfaces 1 Not Configured default Default values are used The default for FXO interfaces is 1 The default for FXS interfaces is 0 0 No PI For IP to Tel calls the device sends a 180 Ringing response to IP after placing a call to a phone FXS or PBX FXO 1 Pl 1 8 Pl 8 For IP to Tel calls if the parameter EnableEarlyMedia is set to 1 the device sends a 183 Session Progress message with SDP immediately after a call is placed to a phone PBX This is used to cut through the voice path before the remote party answers the call This allows the originating party to listen to network Call Progress Tones such as ringback tone or other network announcements Enables the device to send a Re INVITE with a new different SRTP key in the SDP upon receipt of a SIP 181 response call is being forwarded 0 Disable default 1 Enable Note This parameter is applicable only if SRTP is used Defines the maximum number of active SIP dialogs that are not call related i e REGISTER and SUBSCRIBE This parameter is used to control the Registration Subscription rate The valid range is 1 to 5 The default value is 5 Defines the default Release Cause sent to IP for IP to Tel calls when the device initiates a call release and an explicit matching cause for this release
115. Web EMS Prefix to Add Web EMS Suffix to Add Web EMS Number of Digits to Leave Web Presentation EMS Is Presentation Restricted Version 6 0 Description The Hunt Group from where the Tel call is received To denote any Hunt Group leave this field empty Note The value 1 indicates that it is ignored in the rule Destination called telephone number prefix An asterisk represents any number Redirect telephone number prefix An asterisk represents any number Number of digits to remove from the left of the telephone number prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 1234 Number of digits to remove from the right of the telephone number prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 5551 The number or string that you want added to the front of the telephone number For example if you enter 9 and the phone number is 1234 the new number is 91234 The number or string that you want added to the end of the telephone number For example if you enter 00 and the phone number is 1234 the new number is 123400 The number of digits that you want to retain from the right of the phone number Determines whether Caller ID is permitted Not Configured privacy is determined according to the Caller ID table refer to Configuring Caller Display Information on page 138 Allowed sends Caller ID informat
116. a License r 0 m Select the al button to add a new entry and then click Yes at the confirmation prompt a row is added to the table Enter the reguired values Right click the new entry and then from the shortcut menu choose Unlock rows Click Save and then Close Select the IPSec SA tab the IPSec SA screen appears Repeat steps 4 through 7 SIP User s Manual 198 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS 5 6 Provisioning SIP SRTP Crypto Offered Suites This section describes how to configure offered SRTP crypto suites in the SDP gt To configure SRTP crypto offered suites 1 In the MG Tree select the device that you want to configure a graphical representation of the device is displayed in the main pane 2 Open the Authentication amp Security screen Configuration icon gt SIP Protocol Definitions menu gt Authentication amp Security tab Figure 5 7 Authentication amp Security Screen Parameters List EEC Authentication amp Security K a General Info User Name K a Proxy Server Proxy Set Password k k t K W Registration Cnonce Default_Cnonce a Coders Group 0 a DTMF 4 Sup Services Security Enable SIPS Disable v SIPS Require Client Certificate Disable v g Media Security Behavior Preferable K M 0 C E o gof paf i F Authentication amp Security x KI K Kl a Media
117. a currently busy telephone to display the caller ID of the waiting call 0 Caller ID type 2 isn t played 1 Caller ID type 2 is played default 292 Document LTRT 65413 SIP User s Manual Parameter EMS Caller ID Timing Mode AnalogCallerIDTimingMode EMS Bellcore Caller ID Type One Sub Standard BellcoreCallerlIDTypeOneSubSta ndard EMS ETSI Caller ID Type One Sub Standard ETSICallerIDTypeOneSubStanda rd Web Asserted Identity Mode EMS Asserted ID Mode AssertedidMode Version 6 0 6 Configuration Parameters Reference Description Determines when Caller ID is generated 0 Caller ID is generated between the first two rings default 1 The device attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type Notes This parameter is applicable only to FXS interfaces If this parameter is set to 1 and used with distinctive ringing the Caller ID signal doesn t change the distinctive ringing timing For this parameter to take effect a device reset is required Selects the Bellcore Caller ID sub standard 0 Between rings default 1 Not ring related Note For this parameter to take effect a device reset is reguired Selects the ETSI FSK Caller ID Type 1 sub standard FXS only 0 ETSI between rings default 1 ETSI before ring DT AS 2 ETSI before ring RP AS 3 ETSI before
118. advanced Applications gt To view menus in the Navigation tree m On the Navigation bar select the required tab e Configuration refer to Configuration Tab on page 50 e Management refer to Management Tab on page 151 e Status amp Diagnostics refer to Status amp Diagnostics Tab on page 172 Version 6 0 27 March 2010 7a e AudioCodes MediaPack Series 3 1 5 1 gt To navigate to a page 1 Navigate to the required page item by performing the following e Drilling down using the plus signs to expand the menus and submenus e Drilling up using the minus signs to collapse the menus and submenus 2 Select the required page item the page opens in the Work pane Displaying Navigation Tree in Basic and Full View You can view an expanded or reduced Navigation tree display regarding the number of listed menus and submenus This is relevant when using the configuration tabs Configuration Management and Status amp Diagnostics on the Navigation bar The Navigation tree menu can be displayed in one of two views m Basic displays only commonly used menus m Full displays all the menus pertaining to a configuration tab The advantage of the Basic view is that it prevents cluttering the Navigation tree with menus that may not be required Therefore a Basic view allows you to easily locate required menus gt To toggle between Full and Basic view m Select the Basic option located below the
119. and appears listed in the MGs List Note The Pre shared Key string defined in the EMS must be identical to the one that you defined for the device When IPSec is enabled default IPSec IKE parameters are loaded to the device Version 6 0 195 March 2010 A e AudioCodes MediaPack Series 5 4 Configuring Basic SIP Parameters This section describes how to configure the device with basic SIP control protocol parameters using the EMS gt To configure basic SIP parameters 1 In the MG Tree select the device that you want to configure a graphical representation of the device is displayed in the main pane 2 Open the SIP Protocol Definitions frame Configuration icon gt SIP Protocol Definitions menu Parameters List _4 General Info al Proxy Server a Proxy Set Registration 4 Coders Group 0 a DTMF a Sup Services a Authentication a Keypad Features a Media Realm Coders Group 1 r Telephony Profile Figure 5 5 SIP Protocol Definitions Frame WEE General Info 2m Kl Gateway Name K Sip Session Expires K Enable Early Media No Channel Selection Mode CyclicAscending v Fax Used NoFax v Session Expires Method invite m Minimal Session Refresh Value 90 Use SIP URI For Diversion Header tel v Forking Handling Mode Sequential v Offer Unencrypted SR TCP Disable Source Number Preference E E K H a A 2m 3 Select the Cod
120. and the IP address of the FXS device 10 1 10 3 in the field IP Address Figure 9 16 FXO Tel to IP Routing Configuration Dest Phone Prefix Source Phone Prefix Dest IP Address gt E 10 101 103 4 Inthe FXO Settings page refer to Configuring the FXO Parameters on page 151 set the parameter Dialing Mode to Two Stages IsTwoStageDial 1 SIP User s Manual 398 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 5 Configuring Alternative Routing Based on Connectivity and QoS The Alternative Routing feature enables reliable routing of Tel to IP calls when a Proxy isn t used The device periodically checks the availability of connectivity and suitable Quality of Service QoS before routing If the expected quality cannot be achieved an alternative IP route for the prefix phone number is selected The following parameters are used to configure the Alternative Routing mechanism m AltRoutingTel2IPEnable m AltRoutingTel2IPMode E IPConnQoSMaxAllowedPL E IPConnQoSMaxAllowedDelay 9 5 1 Alternative Routing Mechanism When the device routes a Tel to IP call the destination number is compared to the list of prefixes defined in the Tel to IP Routing described in Configuring the Tel to IP Routing on page 126 This table is scanned for the destination number s prefix starting at the top of the table For this reason you must enter the m
121. as ringback busy or fast busy tones One to one mapping occurs between the FXS ports and PBX lines 3 The call disconnects when the phone connected to the FXS goes on hook 9 4 3 2 Dialing from PBX Line or PSTN The procedure below describes how to dial from a PBX line i e from a telephone directly connected to the PBX or from the PSTN to the remote PBX extension i e telephone connected to the FXS interface gt To dial from a telephone directly connected to the PBX or from the PSTN m Dial the PBX subscriber number e g phone number 101 in the same way as if the user s phone was connected directly to the PBX As soon as the PBX rings the FXO device the ring signal is sent to the phone connected to the FXS device Once the phone connected to the FXS device is off hooked the FXO device seizes the PBX line and the voice path is established between the phone and PBX There is one to one mapping between PBX lines and FXS device ports Each PBX line is routed to the same phone connected to the FXS device The call disconnects when the phone connected to the FXS device is on hooked 9 4 3 3 Message Waiting Indication for Remote Extensions The device supports the relaying of Message Waiting Indications MWI for remote extensions and voice mail applications Instead of subscribing to an MWI server to receive notifications of pending messages the FXO device receives subscriptions from the remote FXS device and notifi
122. be listed in the server certificate If the device is operating in HTTPS mode then set the parameter Secured Web Connection HTTPS to HTTP and HTTPS 0 refer to Configuring the General Security Settings on page 78 to ensure you have a method of accessing the device in case the new certificate doesn t work Restore the previous setting after testing the configuration Open the Certificates Signing Request page Configuration tab gt Security Settings menu gt Certificates page item Figure 3 50 Certificates Signing Request Page Certificate Signing Request Subject Name Generate CSR Copy the certificate signing request and send it to your Certification Authority for signing Press the button Generate self signed to create a self signed certificate using the subject name provided above Important this is a lengthy operation during this time the device will be out of service After the operatior complete ave Configuration and reset the device Certificate Files Send Server Certificate file from your computer to the device Browse Send File Send Trusted Root Certificate Store file from your computer to the device Bom Send Private Key file from your computer to the device Browse Send file Note Replacing the private key is not recommended but if it s done it should be over a physically secure network link 73 March 2010 A e AudioCodes MediaPack Series 4 In
123. before the device s FXO interface answers a call by seizing the line The valid range is 0 to 10 The default is 0 When set to 0 the FXO seizes the line after one ring When set to 1 the FXO seizes the line after two rings Notes This parameter is applicable only if automatic dialing is not used If caller ID is enabled and if the number of rings defined by the parameter RingsBeforeCallerID is greater than the number of rings defined by this parameter the greater value is used Determines the dialing mode for IP to Tel FXO calls 0 One Stage One stage dialing 1 Two Stages Two stage dialing default If two stage dialing is enabled the device seizes one of the PSTN PBX lines without performing any dialing connects the remote IP user to the PSTN PBX and all further signaling dialing and Call Progress Tones is performed directly with the PBX without the device s intervention If one stage dialing is enabled the device seizes one of the available lines according to the parameter ChannelSelectMode and dials the destination phone number received in the INVITE message Use the parameter IsWaitForDialTone to specify whether the dialing must start after detection of the dial tone or immediately after seizing the line Note This parameter is applicable only to FXO interfaces Determines whether the device waits for a dial tone before dialing the phone number for IP to Tel FXO calls 0 No Don t wa
124. box appears notifying you of the invalid value In addition the parameter value reverts to its previous value and is highlighted in red as shown in the figure below Figure 3 10 Value Reverts to Previous Valid Value RTP RTCP Settings 000 Basic Parameter List a Invalid Value General Settings s Reverted to Dynamic Jitter Buffer Minimum Delay Previous Valid Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Value Packing Factor Basic RTP Packet Interval Defaut RFC 2833 TX Payload Type 101 RFC 2833 RX Payload Type 101 RFC 2198 Paylosd Type 104 Fax Bypass Payload Type Enable RFC 3389 CN Payload Type 3 1 6 4 Entering Phone Numbers Phone numbers or prefixes that you need to configure throughout the Web interface must be entered only as digits without any other characters For example if you wish to enter the phone number 555 1212 it must be entered as 5551212 without the hyphen If the hyphen is entered the entry is invalid Version 6 0 33 March 2010 7a c tal AudioCodes MediaPack Series 3 1 6 5 Working with Tables The Web interface includes many configuration pages that provide tables for configuring the device Some of these tables provide the following command buttons m Add Index adds an index entry to the table m Duplicate duplicates a selected existing index entry m Compact organizes the index entries in ascending consecutive order m Delete deletes a selected index e
125. busy tone is detected on the device s FXO port default Note This parameter is applicable only to FXO interfaces DisconnectOnBusyTone Polarity Current Reversal for Call Release Analog Interfaces Parameters Web Enable Polarity Reversal Enables the polarity reversal feature for call release EMS Enable Reversal Polarity 3 Disable th larit EnableReversalPolarity aaa Disable the polarity reversal service 1 Enable Enable the polarity reversal service If the polarity reversal service is enabled the FXS interface changes the line polarity on call answer and then changes it back on call release The FXO interface sends a 200 OK response when polarity reversal signal is detected applicable only to one stage dialing and releases a call when a second polarity reversal signal is detected Web EMS Enable Current Disconnect Enables call release upon detection of a Current EnableCurrentDisconnect Disconnect signal 0 Disable Disable the current disconnect service default 1 Enable Enable the current disconnect service If the current disconnect service is enabled The FXO releases a call when a current disconnect signal is detected on its port The FXS interface generates a Current Disconnect Pulse after a call is released from IP The current disconnect duration is determined by the parameter CurrentDisconnectDuration The current SIP User s Manual 316 Document LTRT 654
126. by ensuring that the Use existing configuration check box is marked default e Return the device s configuration settings to factory defaults by not selecting an ini file and by clearing the Use existing configuration check box 7 You can now choose to either e Click Reset the device resets utilizing the new cmp and ini file you loaded up to now as well as utilizing the other auxiliary files e Click Back the Load a cmp file page is opened again e Click Next the next page opens for loading the next consecutive auxiliary file listed in the Wizard 8 For loading the auxiliary files follow the same procedure as for loading the ini file Step 6 9 In the FINISH page complete the upgrade process by clicking Reset the device burns the newly loaded files to flash memory and then resets the device After the device resets the End Process screen appears displaying the burned configuration files refer to the figure below Figure 3 108 End Process Wizard Page Z http 10 13 4 12 EndOfProcess Microsoft Internet Explorer A a CMP Version ID 6 004 003 002 Call Progress Tone File Name usa_tones_13 dat Internet SIP User s Manual 170 Document LTRT 65413 SIP User s Manual 3 Web Based Management 10 Click End Process to close the wizard the Enter Network Password dialog box appears 11 Enter your login user name and password and then click OK a message box appears i
127. by the destination mask columns This gateway address must be on one of the subnets on which the address is configured in the Multiple Interface table 10 8 2 2 4Interface Column This column defines the interface index in the Multiple Interface table from which the gateway address is reached Figure 10 4 Interface Column The Interface Table is a S et ae T RT nm e 0o00 4 Management K 10 32 174 50 16 0000 3 Control A K OE L p 3 1 1 18 105417450 16 0000 7 L 4 5 4 d 2000 1103317450 64 6 vanes The Routing Table Destinatic Prefi m Gatewa nterfac Hop e 201201001 16 10311741 0 1 E S ae The Gateway address resides on the subnet configured in Interface Index 0 at the Interface Table The Next Hop will be accessible via Interface 0 SIP User s Manual 458 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities 10 8 2 2 5Metric Column 10 8 2 3 10 8 2 4 The Metric column must be set to 1 for each routing rule Routing Table Configuration Summary and Guidelines The Routing table configurations must adhere to the following rules m Up to 25 different routing rules may be defined m The user may choose whether to specify Prefix Length or Subnet Mask There is no need to specify both m If both Prefix Length and Subnet Mask are defined the Prefix Length overrides the Subnet Mask m The Gateway IP Addre
128. certain device configurations need to be performed The table below lists the supported event types and subtypes and the corresponding device configurations if required Table 9 3 Supported X Detect Event Types Events Type Subtype Required Configuration CPT SIT NC SITDetectorEnable 1 UserDefinedToneDetectorEnable 1 SIT IC SIT VC Notes Ensure that the CPT file is configured with the SIT RO required tone type Busy On beep detection a SIP INFO message is sent Reorder with type AMD CPT and subtype beep Ringtone The beep detection must be started using regular 9 X detect extension with AMD or CPT reguest beep FAX CED IsFaxUsed 0 or IsFaxUsed 0 and FaxTransportMode 0 modem VxxModemTransportType 3 PTT voice start EnableDSPIPMDetectors 1 voice end The device can detect and report the following Special Information Tones SIT types from the PSTN m SIT NC No Circuit found m SIT IC Operator Intercept m SIT VC Vacant Circuit non registered number m SIT RO Reorder System Busy There are additional three SIT tones that are detected as one of the above SIT tones m The NC SIT tone is detected as NC m The RO SIT tone is detected as RO m The IO SIT tone is detected as VC The device can map these SIT tones to a Q 850 cause and then map them to SIP 5xx 4xx responses using the parameters SITQ850Cause SITQ850CauseForNC SITQ850CauseForlC SITQ850CauseForVC and SITQ850CauseForRO Version 6
129. changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 96 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 4 4 Click the Proxy Set Table gt button to open the Proxy Sets Table page to configure groups of proxy addresses Alternatively you can open this page from the Proxy Sets Table page item refer to Configuring the Proxy Sets Table on page 97 for a description of this page Configuring the Proxy Sets Table The Proxy Sets Table page allows you to define Proxy Sets A Proxy Set is a group of Proxy servers defined by IP address or fully qualified domain name FQDN You can define up to ten Proxy Sets each having a unique ID number and each containing up to five Proxy server addresses For each Proxy server address you can define the transport type i e UDP TCP or TLS In addition Proxy load balancing and redundancy mechanisms can be applied per Proxy Set if a Proxy Set contains more than one Proxy address Proxy Sets can later be assigned to IP Groups of type SERVER only refer to Configuring the IP Groups on page 91 When the device sends an INVITE message to an IP Group it is sent to the IP address or domain name defined for the Proxy Set that is associated with the specific IP Group In other words the Proxy Set represents the destination of the call You can also configure the Proxy Sets table using two complementary ini file table parameters refer to S
130. debugging and for SMDI Parameter DisableRS232 EMS Baud Rate SerialBaudRate Version 6 0 Table 6 16 Serial Parameters Description Enables or disables the device s RS 232 port 0 RS 232 serial port is enabled default 1 RS 232 serial port is disabled The RS 232 serial port can be used to change the networking parameters and view error notification messages For information on establishing a serial communications link with the device refer to the device s Installation Manual Note For this parameter to take effect a device reset is required Determines the value of the RS 232 baud rate The valid values include the following 1200 2400 9600 default 14400 19200 38400 57600 or 115200 Note For this parameter to take effect a device reset is required 229 March 2010 ca AudioCodes Parameter EMS Data SerialData EMS Parity SerialParity EMS Stop SerialStop EMS Flow Control SerialFlowControl 6 3 5 MediaPack Series Description Determines the value of the RS 232 data bit 7 7 bit 8 8 bit default Note For this parameter to take effect a device reset is required Determines the value of the RS 232 polarity 0 None default 1 Odd 2 Even Note For this parameter to take effect a device reset is reguired Determines the value of the RS 232 stop bit 1 1 bit default 2 2 bit Note For thi
131. default 1 Enable The device uses the IP address or domain name defined in the Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 as the Request URI host name in outgoing INVITE messages instead of the value entered in the SIP Group Name field Determines the routing mode after a call redirection i e a 3xx SIP response is received or transfer i e a SIP REFER request is received 0 Standard INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response default 1 Proxy Sends a new INVITE to the Proxy Note 92 Document LTRT 65413 SIP User s Manual 3 Web Based Management Parameter Description Applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0 2 Routing Table Uses the Routing table to locate the destination and then sends a new INVITE to this destination Notes When this parameter is set to 1 and the INVITE sent to the Proxy fails the device re routes the call according to the Standard mode 0 When this parameter is set to 2 and the INVITE fails the device re routes the call according to the Standard mode 0 If DNS resolution fails the device attempts to route the call to the Proxy If routing to the Proxy also fails the Redirect Transfer request is rejected
132. default string AudioCodes product name s w version is used for example User Agent Audiocodes Sip Gateway MediaPack v 6 00 010 006 The maximum string length is 50 characters Note The software version number can t be modified Determines the value of the Owner line o field in outgoing SDP messages The valid range is a string of up to 39 characters The default value is AudiocodesGW For example o AudiocodesGW 1145023829 1145023705 IN IP4 10 33 4 126 Defines the value of the Subject header in outgoing INVITE messages If not specified the Subject header isn t included default The maximum length is up to 50 characters Determines whether the mptime attribute is included in the outgoing SDP 0 None Disabled default 1 PacketCable includes the mptime attribute in the outgoing SDP PacketCable defined format The mptime attribute enables the device to define a separate Packetization period for each negotiated coder in the SDP The mptime attribute is only included if this parameter is enabled even if the remote side includes it in the SDP offer Upon receipt each coder receives its ptime value in the following precedence from mptime attribute from ptime attribute and then from default value Determines whether the ptime attribute is included in the SDP 0 Remove the ptime attribute from SDP 1 Include the ptime attribute in SDP default Determines the dev
133. delay between detection of a Wink and the start of dialing during the establishment of an IP to Tel call for DID lines EnableDIDWink is set to 1 For call transfer the delay after hook flash is generated and dialing begins The valid range in milliseconds is 0 to 20 000 i e 20 seconds The default value is 1 000 i e 1 second Note This parameter is applicable only to FXO interfaces Defines the timeout in seconds for detecting the second ring after the first detected ring If automatic dialing is not used and Caller ID is enabled the device seizes the line after detection of the second ring signal allowing detection of caller ID sent between the first and the second rings If the second ring signal is not received within this timeout the device doesn t initiate a call to IP If automatic dialing is used the device initiates a call to IP when the ringing signal is detected The FXO line is seized only if the remote IP party answers the call If the remote party doesn t answer the call and the second ring signal is not received within this timeout the device releases the IP call This parameter is typically set to between 5 and 8 The default is 8 Notes This parameter is applicable only to FXO interfaces for Tel to IP calls This timeout is calculated from the end of the ring until the start of the next ring For example if the ring cycle is two seconds on and four seconds off the timeout value sh
134. enabled using the AlwaysUseRouteTable parameter even if a proxy server is used the SIP URI host name in the sent INVITE message is obtained from this table Using this feature you can assign a different SIP URI host name for different called and or calling numbers Assign IP Profiles to destination addresses also when a proxy is used Alternative Routing when a proxy isn t used an alternative IP destination can be configured for a specific call type To associate an alternative IP address to a called telephone number prefix assign it with an additional entry with a different IP address or use an FQDN that resolves into two IP addresses The call is sent to the alternative destination when one of the following occurs e Ping to the initial destination is unavailable poor QoS delay or packet loss calculated according to previous calls is detected or a DNS host name is unresolved For detailed information on Alternative Routing refer to Configuring Alternative Routing Based on Connectivity and AoS on page 399 e A release reason defined in the Reasons for Alternative Tel to IP Routing table is received refer to Configuring Reasons for Alternative Routing on page 124 Alternative routing is commonly implemented when there is no response to an INVITE message after INVITE retransmissions The device then issues an internal 408 No Response implicit release reason If this reason is included in the Reasons for Alternat
135. failure response is received for an INVITE request sent by the device 0 Disable default 1 Enable Enables the device to perform SIP re registration upon TCP TLS connection failure 0 Disable default 1 Enable 272 Document LTRT 65413 SIP User s Manual Parameter Web Gateway Registration Name EMS Name GWRegistrationName Web EMS Authentication Mode AuthenticationMode Web Set Out Of Service On Registration Failure EMS Set OOS On Registration Fail OOSOnRegistrationFail UnregistrationMode Version 6 0 6 Configuration Parameters Reference Description Defines the user name that is used in the From and To headers in SIP REGISTER messages If no value is specified default for this parameter the UserName parameter is used instead Note This parameter is applicable only for single registration per device i e AuthenticationMode is set to 1 When the device registers each channel separately i e AuthenticationMode is set to 0 the user name is set to the channel s phone number Determines the device s registration and authentication method 0 Per Endpoint Registration and authentication is performed separately for each endpoint 1 Per Gateway Single registration and authentication for the entire device default 3 Per FXS Registration and authentication for FXS endpoints Typically authentication per endpoint is used for FXS interfaces where e
136. five digits of a phone number to the PBX If for example a company has a PBX with extensions 555 1000 to 555 1999 and a caller dials 555 1234 the local central office CO would forward for example only 234 to the PBX The PBX would then ring extension 234 DID wink enables the originating end to seize the line by going off hook It waits for acknowledgement from the other end before sending digits This serves as an integrity check that identifies a malfunctioning trunk and allows the network to send a re order tone to the calling party The start dial signal is a wink from the PBX to the FXO device The FXO then sends the last four to five DTMF digits of the called number The PBX uses these digits to complete the routing directly to an internal station telephone or equivalent m DID Wink can be used for connection to EIA TIA 464B DID Loop Start lines m Both FXO detection and FXS generation are supported Version 6 0 389 March 2010 7a c tal AudioCodes MediaPack Series 9 4 2 2 9 4 2 2 1 FXO Operations for Tel to IP Calls The FXO device provides the following FXO operating modes for Tel to IP calls m Automatic Dialing refer to Automatic Dialing on page 390 m Collecting Digits Mode refer to Collecting Digits Mode on page 391 m FXO Supplementary Services refer to FXO Supplementary Services on page 391 e Hold Transfer Toward the Tel side e __Hold Transfer Toward the IP side Blind Transfer to t
137. for periodically Connectivity Method querying the connectivity status of a destination IP address EMS Alternative Routing Telephone to IP Connection Method AltRoutingTel2IPConnMethod 1 SIP OPTIONS The remote destination is considered Offline if the latest OPTIONS transaction timed out Any response to an OPTIONS request even if indicating an error brings the connectivity status to online 0 ICMP Ping default Internet Control Message Protocol ICMP ping messages EnableAltMapTel2IP Enables different Tel to IP destination number manipulation rules per routing rule when several up to three Tel to IP routing rules are defined and if alternative routing using release causes is used For example if an INVITE message for a Tel to IP call is returned with a SIP 404 Not Found response the call can be re sent to a different destination number as defined using the parameter NumberMapTel2IP 0 Disable default 1 Enable Web Alt Routing Tel to IP Keep Defines the time interval in seconds between SIP OPTIONS Alive Time Keep Alive messages used for the IP Connectivity application EMS Alternative Routing Keep The valid range is 5 to 2 000 000 The default value is 60 Alive Time AltRoutingTel2IPKeepAliveTime SIP User s Manual 338 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web EMS Alternative Routing Tone Duration ms AltRoutingToneDurati
138. hook flash from the IP side using out of band or RFC 2833 the device sends the hook flash to the Tel side by performing one of the following e Performing a hook flash i e on hook and off hook e Sending a hook flash code defined by the ini file parameter HookFlashCode The PBX may generate a dial tone that is sent to the IP and the IP side may dial digits of a new destination m Blind Transfer to the Tel side A blind transfer is one in which the transferring phone connects the caller to a destination line before ringback begins The ini file parameter LineTransferMode must be set to 1 The blind transfer call process is as follows e FXO receives a REFER request from the IP side e FXO sends a hook flash to the PBX dials the digits that are received in the Refer To header and then drops the line on hook Note that the time between flash to dial is according to the WaitForDialTime parameter e PBX performs the transfer internally m Hold Transfer toward the IP side The FXO device doesn t initiate hold transfer as a response to input from the Tel side If the FXO receives a REFER request with or without replaces it generates a new INVITE according to the Refer To header Version 6 0 391 March 2010 7a c tall AudioCodes MediaPack Series 9 4 23 Call Termination on FXO Devices This section describes the device s call termination capabilities for its FXO interfaces m Calls terminated by a PBX refer to Cal
139. how to configure the device with a pre configured SNMPv3 gt To configure the EMS to operate with a pre configured SNMPv3 system 1 In the MG Tree select the required Region to which the device belongs and then right click the device 2 From the shortcut menu choose Details the MG Information screen appears Figure 5 8 MG Information Screen MG Information General SNMPv2 SNMPv3 SNMP MG Name Device IP Address Engine ID var Security Name snmpy3user1 Description ty P Security Level Authentication amp Privacy Authentication Protocol SHA x IPSec Enabled e Authentication Key P IKE Pre Shared Key Privacy Protocol AES 128 E HTTPS Enabled LJ Privacy Key anansnshe OAM Secure Connection OK Cancel 3 Select the SNMPv3 option configure the SNMP fields and then click OK 4 Open the SNMPv3 Users screen Configuration icon gt Network Frame menu gt SNMPv3 Users tab 5 From the SNMPv3 Users tab s drop down list choose Unit value the SNMPv3 Users table is refreshed with the values that you entered in Step 3 6 Click the Save button the EMS and the device are now synchronized SIP User s Manual 202 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS 5 8 3 Configuring SNMPv3 to Operate with Non Configured SNMPv3 System The procedure below describes how to configure SNMPv3 using the EMS gt Version 6 0 To configure t
140. in SAS Emergency Mode The valid range is 0 Analog to 2 000 000 The default value is 20 Local TCP port used to send receive SIP messages for the SAS application The SIP entities in the local network need to send the registration requests to this port When forwarding the requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5080 Local TLS port used to send receive SIP messages for the SAS application The SIP entities in the local network need to send the registration requests to this port When forwarding the requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5081 Determines whether the device s SAS application adds the SIP Record Route header to SIP requests This ensures that SIP messages traverse the device s SAS agent by including the SAS IP address in the Record Route header 0 Disable default 1 Enable The Record Route header is inserted in a request by a SAS proxy to force future requests in the dialog session to be routed through 310 Document LTRT 65413 SIP User s Manual Parameter Web SAS Proxy Set EMS Proxy Set SASProxySet Web Redundant SAS Proxy Set EMS Redundant Proxy Set RedundantSASProxySet SASEnableContactReplace Web SAS Survivability Mode EMS Survivability Mode SASSurvivabilityMode Version 6 0 6 Config
141. ini file parameters STUNServerPrimarylP and STUNServerSecondaryIP If the primary STUN server isn t available the device attempts to communicate with the secondary server e Define the domain name of the STUN server using the ini file parameter StunServerDomainName The STUN client retrieves all STUN servers with an SRV query to resolve this domain name to an IP address and port sort the server list and use the servers according to the sorted list m Use the ini file parameter NATBindingDefaultTimeout to define the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires STUN only applies to UDP it doesn t support TCP and TLS STUN can t be used when the device is located behind a symmetric NAT Use either the STUN server IP address STUNServerPrimarylP or domain name STUNServerDomainName method with priority to the first one First Incoming Packet Mechanism If the remote device resides behind a NAT device it s possible that the device can activate the RTP RTCP T 38 streams to an invalid IP address UDP port To avoid such cases the device automatically compares the source address of the incoming RTP RTCP T 38 stream with the IP address and UDP port of the remote device If the two are not identical the transmitter modifies the sending address to correspond with the address of the incoming stream The RTP RTCP and T 38 can thus have independent destination IP add
142. is 0 i e the update at fixed intervals mechanism is disabled Note For this parameter to take effect a device reset is required AutoUpdatePredefinedTime Schedules an automatic update to a user defined time of the day The format of this parameter is HH MM where HH depicts the hour and MM the minutes for example 20 18 Notes For this parameter to take effect a device reset is required The actual update time is randomized by five minutes to reduce the load on the Web servers ResetNow Invokes an immediate device reset This option can be used to activate offline i e not on the fly parameters that are loaded using the parameter IniFileUrl 0 The immediate restart mechanism is disabled default 1 The device immediately resets after an ini file with this parameter set to 1 is loaded SIP User s Manual 362 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Software Configuration File URL Path for Automatic Update Parameters CmpFileURL Specifies the name of the cmp file and the path to the server IP address or FQDN from where the device loads a new cmp file and updates itself The cmp file can be loaded using HTTP HTTPS FTP FTPS or NFS For example http 192 168 0 1 filename Notes For this parameter to take effect a device reset is required When this parameter is configured the device always loads the cmp file afte
143. is not found The default release cause is NO ROUTE TO DESTINATION 3 Other common values include NO CIRCUIT AVAILABLE 34 DESTINATION OUT OF ORDER 27 etc Notes The default release cause is described in the Q 931 notation and is translated to corresponding SIP 40x or 50x values e g 3 to SIP 404 and 34 to SIP 503 Foran explanation on mapping PSTN release causes to SIP responses refer to Mapping PSTN Release Cause to SIP Response on page 432 For a list of SIP responses Q 931 release cause mapping refer to Release Reason Mapping Modifies the called number for numbers received with Microsoft s proprietary ext xxx parameter in the SIP INVITE URI user part Microsoft Office Communications Server sometimes uses this proprietary parameter to indicate the extension number of the called party 0 Disable default 1 Enable For example if a calling party makes a call to telephone number 622125519100 Ext 104 the device receives the SIP INVITE 258 Document LTRT 65413 SIP User s Manual Parameter EMS Use SIP URI For Diversion Header UseSIPURIForDiversionHeader TimeoutBetween100And18x Web Comfort Noise Generation Negotiation EMS Comfort Noise Generation ComfortNoiseNegotiation Web EMS First Call Ringback Tone ID FirstCallIRBTId Web Reanswer Time EMS Regret Time RegretTime Version 6 0 6 Configuration Parameters Reference Description from Micros
144. is not required i e leave the field empty The IP Group from where the IP to IP call originated Typically this IP Group of an incoming INVITE is determined classified using the IP to Hunt Group Routing Table If not used i e any IP Group simply leave the field empty Notes The value 1 indicates that it is ignored in the rule This parameter is available only in the Source Phone Number Manipulation Table for Tel gt IP Calls and Destination Phone Number Manipulation Table for Tel gt IP Calls pages If this Source IP Group has a Serving IP Group then all calls originating from this Source IP Group is sent to the Serving IP Group In this scenario this table is used only if the parameter PreferRouteTable is set to 1 Destination called telephone number prefix An asterisk represents any number Source calling telephone number prefix An asterisk represents any number Source IP address of the caller obtained from the Contact header in the INVITE message Notes This parameter is applicable only to the Number Manipulation tables for IP to Tel calls The source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all IP addresses between 10 8 8 10 to 10 8 8 99 The source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all IP addresses between 10 8
145. later e Mozilla Firefox version 2 5 or later m Required minimum screen resolution 1024 x 768 pixels or 1280 x 1024 pixels Note Your Web browser must be JavaScript enabled to access the Web interface Version 6 0 23 March 2010 A e AudioCodes MediaPack Series 3 1 2 Accessing the Web Interface The Web interface can be opened using any standard Web browser refer to Computer Requirements on page 23 When initially accessing the Web interface use the default user name Admin and password Admin For changing the login user name and password refer to Configuring the Web User Accounts on page 66 Note For assigning an IP address to the device refer to the device s Installation Manual gt To access the Web interface 1 Open a standard Web browser application 2 In the Web browser s Uniform Resource Locator URL field specify the device s IP address e g http 10 1 10 10 the Web interface s Enter Network Password dialog box appears as shown in the figure below Figure 3 1 Enter Network Password Screen Enter Network Password This secure Web Site at 10 33 4 128 requires you to log on Please type the User Name and Password that you use for Realm User Name v Password IV Save this password in your password list Cancel 3 In the User Name and Password fields enter the case sensitive user name and password 4 Click the OK button the We
146. located in the same folder as the ini file For a detailed description on BootP refer to the Product Reference Manual SIP User s Manual 42 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 8 6 Deleting a Scenario You can delete the Scenario by using the Delete Scenario File button as described in the procedure below gt To delete the Scenario 1 On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm Figure 3 19 Scenario Loading Message Box z Microsoft Internet Explorer A Loading Scenario PBX Interoperability 2 Click OK the Scenario mode appears in the Navigation tree 3 Click the Delete Scenario File button a message box appears requesting confirmation for deletion Figure 3 20 Message Box for Confirming Scenario Deletion Microsoft Internet Explorer 2 J This operation will delete the current scenario file are you sure 4 Click OK the Scenario is deleted and the Scenario mode closes Note You can also delete a Scenario using the following alternative methods e Loading an empty dat file refer to Loading a Scenario to the Device on page 42 Loading an ini file with the ScenarioFileName parameter set to no value i e ScenarioFileName 3 1 8 7 Exiting Scenario Mode When you want to close the Scenario mode after using it for device configuration follow the procedure below Version 6 0 43 March 2010
147. neds tat nebavi budo ee aa dete indee plane dats 465 Table 12 1 MediaPack Technical Specifications c ccccecceeeecececeeeeeeeeeeeaeeeceeeeeeeeeeeeneaeeeeeeeeeeeees 467 SIP User s Manual 14 Document LTRT 65413 SIP User s Manual Notices Notice This document describes the AudioCodes MediaPack series Voice over IP VoIP gateways Information contained in this document is believed to be accurate and reliable at the time of printing However due to ongoing product improvements and revisions AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions Before consulting this document check the corresponding Release Notes regarding feature preconditions and or specific support in this release In cases where there are discrepancies between this document and the Release Notes the information in the Release Notes supersedes that in this document Updates to this document and other documents can be viewed by registered customers at http www audiocodes com downloads Copyright 2010 AudioCodes Ltd All rights reserved This document is subject to change without notice Date Published March 14 2010 Trademarks AudioCodes AC AudioCoded Ardito CTI2 CTI CTI Squared HD VoIP HD VoIP Sounds Better InTouch IPmedia Mediant MediaPack NetCoder Netrake Nuera Open Solutions Network OSN Stretto TrunkPack VMAS VoicePacketizer VolPerfect
148. not established The valid range is 1 to the maximum number of supported channels The default value is the maximum available channels i e no restriction on the maximum number of calls PRACK Provisional Acknowledgment mechanism mode for SIP 1xx reliable responses 0 Disable 1 Supported default 2 Required Notes The Supported and Required headers contain the 100rel tag The device sends PRACK messages if 180 183 responses are received with 100rel in the Supported or Required headers Enables the device to send a 183 Session Progress response with SDP instead of a 180 Ringing allowing the media stream to be established prior to the answering of the call 0 Disable Early Media is disabled default 1 Enable Enables Early Media Note that to send a 183 response you must also set the parameter ProgressIndicator2IP to 1 If it is equal to 0 180 Ringing response is sent 245 March 2010 ca AudioCodes Parameter Web 183 Message Behavior EMS SIP 183 Behaviour SIP183Behaviour Web Session Expires Time EMS Sip Session Expires SIPSessionExpires Web Minimum Session Expires EMS Minimal Session Refresh Value MinSE Web EMS Session Expires Method SessionExpiresMethod RemoveToTaglInFailureRespon se EnableRTCPAttribute EMS Options User Part OPTIONSUserPart Web Fax Signaling Method EMS Fax Used IsFaxUsed SIP User s Manual MediaPack Series
149. of these reasons the device attempts to locate an alternative Hunt Group for the call in the IP to Hunt Group Routing Table The format of this parameter is as follows AltRouteCauselP2Tel FORMAT AltRouteCauselP2Tel Index AltRouteCauselP2Tel ReleaseCause AltRouteCauselP2Tel For example AltRouteCauselP2Tel 0 3 No Route to Destination AltRouteCauselP2Tel 1 1 Unallocated Number AltRouteCauselP2Tel 2 17 Busy Here Notes This parameter can include up to 5 indices This table can be used for example in scenarios where the destination is busy and the Release Reason 17 is issued or for other call releases that issue the default Release Reason 3 The device also plays a tone to the endpoint whenever an alternative route is used This tone is played for a user defined time configured by the parameter AltRoutingToneDuration Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Forward On Busy Trunk Destination ForwardOnBusyTrunkDest This ini file table parameter configures the Forward On Busy Trunk Destination table This table allows you to define an alternative IP destination IP address per Hunt Group for IP to Tel calls The IP to Tel call is forwarded to this IP destination using 3xx response if the following an FXO FXS Hunt Group has no free channels This feature can be used for example to forward the call to anoth
150. off m Echo Canceller Non Linear Processor Mode off SIP User s Manual 402 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 6 2 4 m Dynamic Jitter Buffer Minimum Delay 40 m Dynamic Jitter Buffer Optimization Factor 13 When the device initiates a fax session using G 711 a gpmd attribute is added to the SDP according to the following format m For G 711A law a gpmd 0 vbd yes ecan on m For G 711 u law a gpmd 8 vbd yes ecan on In this mode the parameter FaxTransportMode is ignored and automatically set to transparent To configure fax fallback mode set IsFaxUsed to 3 Fax Modem Bypass Mode In this proprietary mode when fax or modem signals are detected the channel automatically switches from the current voice coder to a high bit rate coder according to the parameter FaxModemBypassCoderType In addition the channel is automatically reconfigured with the following fax modem adaptations m Disables silence suppression m Enables echo cancellation for fax m Disables echo cancellation for modem m Performs certain jitter buffering optimizations The network packets generated and received during the bypass period are regular voice RTP packets per the selected bypass coder but with a different RTP payload type according to the parameters FaxBypassPayloadType and ModemBypassPayloadType During the bypass period the coder uses the packing factor which is defined by the parameter
151. or re INVITE messages m Emergency The SAS agent switches to this mode if it detects from the keep alive responses that the connection with the Proxy is lost This can occur due to Proxy server failure or WAN problems In this mode when the connection with the Proxy server is down the SAS agent handles all internal calls within the enterprise In the case of outgoing calls the SAS agent forwards these to a local VoIP gateway this can be the device itself or a separate analog or digital gateway For PSTN fallback the local VoIP gateway should be equipped with analog FXO lines for PSTN connectivity In this way the enterprise preserves its capability for internal and outgoing calls The call routing rules for SAS is configured in the IP2IP Routing Table page refer to Configuring the IP2IP Routing Table SAS on page 146 This table provides enhanced call routing capabilities such as built in ENUM queries and redundant SAS proxy server load balancing for routing received SIP INVITE messages When SAS receives a SIP INVITE request from a Proxy server the following routing logic is performed a Sends the request according to rules configured in the IP2IP Routing table b If no matching routing rule exists the device sends the request according to its SAS registration database c If no routing rule is located in the database the device sends the request according to the Request URI header Version 6 0 381 March 2010 7a
152. page 161 Version 6 0 63 March 2010 7a K tal AudioCodes MediaPack Series 3 3 2 4 Configuring the General Media Settings The General Media Settings page allows you to configure various media parameters For a detailed description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt To configure general media parameters 1 Open the General Media Settings page Configuration tab gt Media Settings menu gt General Media Settings page item Figure 3 43 General Media Settings Page w General Settings Enable Continuity Tones Disable 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 2 5 Configuring the Analog Settings The Analog Settings page allows you to configure various analog parameters For a detailed description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 This page also selects the type USA or Europe of FXS and or FXO coefficient information The FXS coefficient contains the analog telephony interface characteristics such as DC and AC impedance feeding current and ringing voltage gt To configure the analog parameters 1 Open the Analog Settings page Configuration tab gt Media Settings menu gt Analog Settings page item Figure 3 44 Ana
153. parameters appearing on this page refer to Configuration Parameters Reference on page 207 SIP User s Manual 88 Document LTRT 65413 SIP User s Manual 2 3 4 Version 6 0 To configure the general SIP protocol parameters Open the SIP General Parameters page Configuration 3 Web Based Management tab gt Protocol Configuration menu gt Protocol Definition submenu gt SIP General Parameters page item Figure 3 58 SIP General Parameters Page w SIP General NAT IP Address PRACK Mode Channel Select Mode Enable Early Media 183 Message Behavior Session Expires Time Minimum Session Expires Session Expires Method Asserted Identity Mode Fax Signaling Method Detect Fax on Answer Tone SIP Transport Type SIP UDP Local Port SIP TCP Local Port SIP TLS Local Port Enable SIPS Enable TCP Connection Reuse TCP Timeout SIP Destination Port Use user phone in SIP URL Use user phone in From Header Use Tel URI for Asserted Identity Tel to IP No Answer Timeout Enable Remote Party ID Add Number Plan and Type to RPI Header Enable History Info Header Use Source Number as Display Name Use Display Name as Source Number Enable Contact Restriction Play Ringback Tone to IP Play Ringback Tone to Tel Use Tarp information Enable GRUU User Agent Information SDP Session Owner Subject Multiple Packetization Time Format Enable Semi Attended Transfer 3xx Behavior Enable P Charging Ve
154. published in SDP for RTP and for T38 must be different Therefore set the the parameter T38UseRTPPort to 0 Web EMS T 38 Max Datagram Size T38MaxDatagramSize Version 6 0 Defines the maximum size of a T 38 datagram that the device can receive This value is included in the outgoing SDP when T 38 is used The valid range is 122 to 1 024 The default value is 122 27T March 2010 ca AudioCodes Parameter Web EMS T38 Fax Max Buffer T38FaxMaxBufferSize Web EMS Enable Fax Re Routing EnableFaxReRouting Web EMS Fax CNG Mode FaxCNGMode Web Detect Fax on Answer Tone EMS Enables Detection of FAX on Answer Tone DetFaxOnAnswerTone SIP User s Manual MediaPack Series Description Defines the maximum size in bytes of the device s T 38 buffer This value is included in the outgoing SDP when T 38 is used for fax relay over IP The valid range is 100 to 1 024 The default value is 1 024 Enables or disables re routing of Tel to IP calls that are identified as fax calls 0 Disable Disabled default 1 Enable Enabled If a CNG tone is detected on the Tel side of a Tel to IP call the prefix FAX is appended to the destination number before routing and manipulations A value of FAX entered as the destination number in the Tel to IP Routing is then used to route the call and the destination number manipulation mechanism is used to remove the FAX prefix if reguired If the initi
155. receiving a request with History Info the UAS checks the policy in the request If a session header or history policy tag is found the final response is sent without History Info otherwise it is copied from the request Determines whether the SIP tgrp parameter is used This SIP parameter specifies the Hunt Group to which the call belongs according to RFC 4904 For example the SIP message below indicates that the call belongs to Hunt Group ID 1 INVITE sip 16305550100 tgrp 1 trunk context example com 10 1 0 3 user phone SIP 2 0 0 Disable default The tgrp parameter isn t used 1 Send Only The Hunt Group number is added to the tgrp parameter value in the Contact header of outgoing SIP messages If a Hunt Group number is not associated with the call the tgrp parameter isn t included If a tgrp value is specified in incoming messages it is ignored 2 Send and Receive The functionality of outgoing SIP 250 Document LTRT 65413 SIP User s Manual Parameter Web EMS TGRP Routing Precedence TGRProutingPrecedence UseBroadsoftDTG Version 6 0 6 Configuration Parameters Reference Description messages is identical to the functionality described in option 1 In addition for incoming SIP INVITEs if the Request URI includes a tgrp parameter the device routes the call according to that value if possible The Contact header in the outgoing SIP INVITE Tel to IP call
156. required The Native VLAN ID is the same VLAN ID as the AudioCodes Management interface index 0 One routing rule is required to allow remote management from a host in 176 85 49 0 24 Table 10 14 Routing Table Example 3 Destination Prefix Length Subnet Mask Gateway Interface Metric 176 85 49 0 24 192 168 0 1 0 1 All other parameters are set to their respective default values The ini file matching this configuration can be written as follows Interface Table Configuration InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes InterfaceTable InterfaceMode InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway InterfaceTable VlanID InterfaceTable InterfaceName InterfaceTable 0 0 10 192 168 8514 16 0 0 0 0 1 Mgmt InterfaceTable 1 5 iO 20020085 14 Aa A GO 200 85 i1 AWM CntrlMedia1 InterfaceTable 2 5 10 200 200 86 14 24 0 0 0 0 202 CntrlMedia2 InterfaceTable VLAN related parameters VlanMode 1 VlanNativeVlanId 1 Routing Table Configuration RoutingTableDestinationsColumn 176 85 49 0 RoutingTableDestinationPrefixLensColumn 24 RoutingTableGatewaysColumn 192 168 0 1 RoutingTableInterfacesColumn 0 RoutingTableHopsCountColumn 1 Version 6 0 463 March 2010 A c tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 464 Document LTRT 65413 SIP User s Manual 11 SIP Software Package 1
157. ring LR DT AS 4 ETSI not ring related DT AS 5 ETSI not ring related RP AS 6 ETSI not ring related LR DT AS Note For this parameter to take effect a device reset is reguired Determines whether P Asserted Identity or P Preferred Identity is used in the generated INVITE reguest for Caller ID or privacy 0 Disabled None default 1 Adding PAsserted Identity 2 Adding PPreferred Identity This parameter determines the header P Asserted Identity or P Preferred Identity used in the generated INVITE request The header also depends on the calling Privacy allowed or restricted These headers are used to present the originating party s Caller ID The Caller ID is composed of a Calling Number and optionally a Calling Name These headers are used together with the Privacy header If Caller ID is restricted i e P Asserted Identity is not sent the Privacy header includes the value id Privacy id Otherwise for allowed Caller ID Privacy none is used If Caller ID is 293 March 2010 ca AudioCodes Parameter Web Caller ID Transport Type EMS Transport Type CallerlIDTransportType MediaPack Series Description restricted received from Tel or configured in the device the From header is set to lt anonymous anonymous invalid gt Determines the device s behavior for Caller ID detection 0 Disable The caller ID signal is not detected DTMF digits remain in the
158. s Manual 3 Web Based Management 3 3 4 8 7 gt To configure the Internal SRV table 1 Open the Internal SRV Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Internal SRV Table page item Figure 3 82 Internal SRV Table Page v v v 2 In the Domain Name field enter the host name to be translated You can enter a string of up to 31 characters long 3 From the Transport Type drop down list select a transport type In the DNS Name 1 field enter the first DNS A Record to which the host name is translated In the Priority Weight and Port fields enter the relevant values Repeat steps 4 through 5 for the second and third DNS names if reguired Repeat steps 2 through 6 for each entry Click the Submit button to save your changes 7 74 0 0 To save the changes so they are available after a hardware reset or power fail refer to Saving Configuration on page 161 Configuring Call Forward upon Busy Trunk The Forward on Busy Trunk Destination page allows you to configure forwarding of IP to Tel calls to a different alternative IP destination using SIP 3xx response upon the following scenario m If an unavailable FXS FXO Hunt Group exists This feature can be used for example to forward the call to another FXS FXO device The alternative destination i e IP address port and transport type is configured per Hunt
159. same tone that exists in the CPT file and is played instead of it Note The PRT are used only for generation of tones Detection of tones is performed according to the CPT file The PRT is a dat file containing a set of prerecorded tones that can be played by the device Up to 40 tones totaling approximately 10 minutes can be stored in a single PRT file on the device s flash memory The prerecorded tones are prepared offline using standard recording utilities such as CoolEdit and combined into a single file using the DConvert utility refer to the Product Reference Manual The raw data files must be recorded with the following characteristics m Coders G 711 A law or G 711 u law m Rate 8 kHz m Resolution 8 bit m Channels mono SIP User s Manual 372 Document LTRT 65413 SIP User s Manual 8 Auxiliary Configuration Files 8 3 Once created the PRT file can then be loaded to the device using AudioCodes BootP TFTP utility or the Web interface refer to Loading Auxiliary Files on page 163 The prerecorded tones are played repeatedly This allows you to record only part of the tone and then play the tone for the full duration For example if a tone has a cadence of 2 seconds on and 4 seconds off the recorded file should contain only these 6 seconds The PRT module repeatedly plays this cadence for the configured duration Similarly a continuous tone can be played by repeating only part of it Dial Pla
160. secondary dial tone i e stutter tone to the FXS line and then starts collecting the subsequently dialed digits from the FXS line The valid range is a one character string The default is an empty string Notes You can enable the device to add this string as the prefix to the collected and sent digits using the parameter AddPrefix2ExtLine This parameter is applicable only to FXS interfaces AddPrefix2ExtLine Determines whether the prefix string for accessing an external line defined by the parameter Prefix2ExtLine is added to the dialed number as the prefix and together sent to the IP destination Tel to IP calls 0 Disable default 1 Enable For example if this parameter is enabled and the prefix string for the external line is defined as 9 using the parameter Prefix2ExtLine and the FXS user wants to make a call to destination 123 the device collects and sends all the dialed digits including the prefix string as 9123 to the IP destination number Note This parameter is applicable only to FXS interfaces SIP User s Manual 324 Document LTRT 65413 SIP User s Manual Parameter Hook Flash Parameters Web Flash Keys Sequence Style FlashKeysSequenceStyle Web Flash Keys Sequence Timeout FlashKeysSequenceTimeout 6 Configuration Parameters Reference Description Hook flash keys sequence style 0 0 Flash hook default only the phone s Flash button is use
161. service class content PremiumServiceClassControlDiffServ control traffic Version 6 0 453 March 2010 ca AudioCodes MediaPack Series Parameter Description Sets the DiffServ for the Gold service class content GoldServiceClassDiffServ streaming traffic Sets the DiffServ for the Bronze service class content BronzeServiceClassDiffServ OAMP traffic The mapping of an application to its CoS and traffic type is shown in the table below Table 10 6 Traffic Network Types and Priority Application Traffic Network Types Class of Service Priority Debugging interface Management Bronze Telnet Management Bronze DHCP Management Network Web server HTTP Management Bronze SNMP GET SET Management Bronze Web server HTTPS Management Bronze IPSec IKE Determined by the service Determined by the service RTP traffic Media Premium media RTCP traffic Media Premium media T 38 traffic Media Premium media SIP Control Premium control SIP over TLS SIPS Control Premium control Syslog Management Bronze ICMP Management oo by the initiator of the ARP listener allies by the initiator of the Network SNMP Traps Management Bronze DNS client DNS EnableDNSasOAM Network Depends on traffic type NTP NTP EnableNTPasOAM Control Premium control Management Bronze NFS NFSServers_VlanType in the Gold NFSServers table 10 8 1 3 5Applications with Assignable Application Type Some applications can be associated with different applicati
162. sip 200 tgrp 7 trunk context example com 10 33 2 68 user phone SIP 2 0 Notes For enabling routing based on the tgrp parameter the UseSIPTgrp parameter must be set to 2 For IP to Tel routing based on the dtg parameter instead of the tgrp parameter use the parameter UseBroadsoftDTG Determines whether the device uses the dtg parameter for routing IP to Tel calls to a specific Hunt Group 0 Disable default 1 Enable When this parameter is enabled if the Request URI in the received SIP INVITE includes the dtg parameter the device routes the call to the Hunt Group according to its value This parameter is used instead of the tgrp trunk context parameters The dtg parameter appears in the INVITE Request URI and in the To header For example the received SIP message below routes the call to Hunt Group ID 56 251 March 2010 ca AudioCodes Parameter Web EMS Enable GRUU EnableGRUU EMS Is CISCO Sce Mode IsCiscoSCEMode SIP User s Manual MediaPack Series Description INVITE sip 123456 192 168 1 2 dtg 56 user phone SIP 2 0 Note If the Hunt Group is not found based on the dtg parameter the IP to Hunt Group Routing Table is used instead for routing the call to the appropriate Hunt Group Determines whether the Globally Routable User Agent URIs GRUU mechanism is used 0 Disable default 1 Enable The device obtains a GRUU by generating a normal REGISTER req
163. space Upon device start up this table is parsed and passes comprehensive validation tests If any errors occur during this validation phase the device sends an error message to the Syslog server and falls back to a safe mode using a single IPv4 interface and without VLANs Therefore check the Syslog for any error messages When booting using BootP DHCP protocols an IP address is obtained from the server This address is used as the OAMP address for this session overriding the address configured using the InterfaceTable The address specified for OAMP applications in this becomes available when booting from flash again This enables the device to work with a temporary address for initial management and configuration while retaining the address to be used for deployment For configuring additional routing rules for other interfaces use the Tel to IP Routing To configure multiple IP interfaces in the Web interface and for a detailed description of the table s parameters refer to Configuring the Multiple Interface Table on page 50 Fora description of configuring ini file table parameters refer to Configuring ini File Table Parameters on page 186 The device s source IP address in the operations administration maintenance and provisioning OAMP network The default value is 0 0 0 0 Note For this parameter to take effect a device reset is required The device s subnet mask in the OAMP networ
164. standard Web browser e g Microsoft Internet Explorer Access to the Web interface is controlled by various security mechanisms such as login user name and password read write privileges and limiting access to specific IP addresses This section includes full parameter descriptions for the Web interface configuration tables only For descriptions of individual parameters refer to Configuration Parameters Reference on page 207 The Web interface allows you to configure most of the device s parameters Those parameters that are not available in the Web interface can be configured using the ini file Throughout this section parameters enclosed in square brackets depict the corresponding ini file parameters Some Web interface pages are Software Upgrade Key dependant These pages appear only if the installed Software Upgrade Key supports the features related to these pages For viewing your Software Upgrade Key refer to Upgrading the Software Upgrade Key on page 165 3 1 Getting Acquainted with the Web Interface This section describes the Web interface with regards to its graphical user interface GUI and basic functionality 3 1 1 Computer Requirements To use the device s Web interface the following is required m A connection to the Internet network World Wide Web m A network connection to the device s Web interface m One of the following Web browsers e Microsoft Internet Explorer version 6 0 or
165. table refer to Configuring Hunt Group Settings on page 85 or using the TrunkGroupSettings ini file parameter SIP User s Manual 182 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 5 2 5 Viewing IP Connectivity The IP Connectivity page displays online read only network diagnostic connectivity information on all destination IP addresses configured in the Tel to IP Routing page refer to Configuring the Tel to IP Routing on page 126 This information is available only if the parameter Enable Alt Routing Tel to IP AltRoutingTel2IPMode refer to Configuring Routing General Parameters on page 125 is set to 1 Enable or 2 Status Only The information in columns Quality Status and Quality Info per IP address is reset if two minutes elapse without a call to that destination gt To view the IP connectivity information 1 In the Routing General Parameters page set the parameter Enable Alt Routing Tel to IP or ini file parameter AltRoutingTel2IPEnable to Enable 1 or Status Only 2 2 Open the IP Connectivity page Status amp Diagnostics tab gt Gateway Statistics menu gt IP Connectivity page item Figure 3 118 IP Connectivity Page Connectivity Connectivity Quality IP Address Host Name Method Status Status Quality Info DNS Status Unused Unused Unused Unused Unused Unused Unused Unused Unused 10 Unused 1 2 3 4 E 6 7 8 9
166. the Signal On Time parameter of the continuous tone must have a value that is greater than the Signal On Time parameter of the cadence tone Otherwise the continuous tone is detected instead of the cadence tone The tones frequency must differ by at least 40 Hz between defined tones For example to configure the dial tone to 440 Hz only enter the following text NUMBER OF CALL PROGRESS TONES Number of Call Progress Tones 1 Dial Tone CALL PROGRESS TONE 0 Tone Type 1 Tone Form 1 continuous Low Freq Hz 440 High Freq Hz 0 Low Freq Level dBm 10 10 dBm High Freq Level dBm 32 use 32 only if a single tone is required Version 6 0 369 March 2010 A tall AudioCodes MediaPack Series First Signal On Time 10msec 300 the dial tone is detected after 3 sec First Signal Off Time 10msec 0 Second Signal On Time 10msec 0 Second Signal Off Time 10msec 0 Distinctive Ringing Distinctive Ringing is applicable only to FXS interfaces Using the Distinctive Ringing section of the Call Progress Tones auxiliary file you can create up to 16 Distinctive Ringing patterns Each ringing pattern configures the ringing tone frequency and up to four ringing cadences The same ringing frequency is used for all the ringing pattern cadences The ringing frequency can be configured in the range of 10 to 200 Hz with a 5 Hz resolution Each of the ringing pattern cadences is specif
167. the FXO line to restore connection to the original call 3 Supervised PBX Supervised transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX and then dials the digits that are received in the Refer To header The FXO waits for connection of the transfer call and if speech is detected e g hello within approximately 2 seconds the device completes the call transfer by releasing the line otherwise the transfer is cancelled the device sends a SIP NOTIFY message with a failure reason in the NOTIFY body such as 486 if busy tone detected and generates an additional hook flash towards the FXO line to restore connection to the original call Enables Simplified Message Desk Interface SMDI interface on the device 0 Disable Normal serial default 1 Enable Bellcore 2 Ericsson MD 110 3 NEC ICS Notes For this parameter to take effect a device reset is required When the RS 232 connection is used for SMDI messages Serial SMDI it cannot be used for other applications for example to access the Command Line Interface CLI Determines the time in msec that the device waits for an SMDI Call Status message before or after a Setup message is received This parameter synchronizes the SMDI and analog CAS interfaces If the timeout expires and only an SMDI message is received the SMDI message is dropped If the timeout expires and only a Setup messa
168. the available pattern syntaxes refer to the CPE Configuration Guide for Voice Mail Web Forward on Busy Digit Pattern Internal EMS Digit Pattern Forward On Busy DigitPatternForwardOnBusy Web Forward on No Answer Digit Pattern Internal EMS Digit Pattern Forward On No Answer DigitPatternForwardOnNoAnswer Web Forward on Do Not Disturb Digit Pattern Internal EMS Digit Pattern Forward On DND DigitPatternForwardOnDND Web Forward on No Reason Digit Pattern Internal EMS Digit Pattern Forward No Reason DigitPatternForwardNoReason Web Forward on Busy Digit Pattern External EMS VM Digit Pattern On Busy External DigitPatternForwardOnBusyExt Web Forward on No Answer Digit Pattern External EMS VM Digit Pattern On No Answer Ext DigitPatternForwardOnNoAnswerExt SIP User s Manual Determines the digit pattern used by the PBX to indicate call forward on busy when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on no answer when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on do not disturb when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by th
169. the device doesn t detect a broken RTP connection During a call if the source IP address from where the RTP packets are received is changed without notifying the device the device filters these RTP packets To overcome this set the parameter DisconnectOnBrokenConnection to 0 the device doesn t detect RTP packets arriving from the original source IP address and switches after 300 msec to the RTP packets arriving from the new source IP address The time period in 100 msec units after which a call is disconnected if an RTP packet is not received The valid range is 1 to 1 000 The default value is 100 i e 10 seconds Notes This parameter is applicable only if the parameter DisconnectOnBrokenConnection is set to 1 Currently this feature functions only if Silence Suppression is disabled Determines whether calls are disconnected after detection of silence 1 Yes The device disconnects calls in which silence occurs in both call directions for more than a user defined time 0 No Call is not disconnected when silence is detected default The silence duration can be configured by the FarEndDisconnectSilencePeriod parameter default 120 Note To activate this feature set the parameters EnableSilenceCompression and FarEndDisconnectSilenceMethod to 1 Duration of the silence period in seconds after which the call is disconnected The range is 10 to 28 800 i e 8 hours The default is 1
170. the inbound IP routing rule according to the table below 4 Click the Submit button to save your changes 5 To save the changes so they are available after a power failure refer to Saving Configuration on page 161 Parameter IP to Tel Routing Mode RouteModelP2Tel Dest Host Prefix Source Host Prefix Dest Phone Prefix Source Phone Prefix SIP User s Manual Table 3 22 IP to Tel Routing Table Description Description Determines whether to route the incoming IP calls before or after manipulation of destination number configured in Configuring the Number Manipulation Tables on page 115 0 Route calls before manipulation Incoming IP calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation Incoming IP calls are routed after the number manipulation rules are applied The Request URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty Note The asterisk wildcard can be used to depict any prefix The From URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty Notes The asterisk wildcard can be used to depict any prefix If the P Asserted Identity header is present in the incoming INVITE message then the value of this parameter is compared to the P Asserted Identity URI host name and not the From header
171. there is no response and if DHCP is disabled the device boots from flash It then attempts to communicate with the DHCP server to SIP User s Manual 220 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description renew the lease Note For this parameter to take effect a device reset is required DHCPRequestTFTPParams Determines whether the device includes DHCP options 66 and 67 in DHCP Option 55 Parameter Request List for requesting the DHCP server for TFTP provisioning parameters 0 Disable default 1 Enable Note For this parameter to take effect a device reset is required 6 1 9 NTP and Daylight Saving Time Parameters The Network Time Protocol NTP and daylight saving time parameters are described in the table below Table 6 9 NTP and Daylight Saving Time Parameters Parameter Description NTP Parameters Note For detailed information on Network Time Protocol NTP refer to Simple Network Time Protocol Support on page 447 Web NTP Server IP Address The IP address in dotted decimal notation of the NTP server EMS Server IP Address The default IP address is 0 0 0 0 i e internal NTP client is NTPServerlP disabled Web NTP UTC Offset Defines the Universal Time Coordinate UTC offset in seconds EMS UTC Offset from the NTP server NTPServerUTCOffset The default offset is 0 The offset range is 43200 to 43200 Web NTP Update Interval Define
172. to 3 0 Group 1 768 Bits DH 786 Bit 41 Group 2 1024 Bits default DH 1024 Bit If no proposals are defined the default settings shown in the following table are applied Table 3 11 Default IPSec IKE Proposals Proposal Encryption Authentication DH Group Proposal 0 3DES SHA1 Group 2 1024 bit Proposal 1 3DES MD5 Group 2 1024 bit Proposal 2 3DES SHA1 Group 1 786 bit Proposal 3 3DES MD5 Group 1 786 bit 3 3 3 8 Configuring the IP Security Associations Table The IP Security Associations Table page allows you to configure up to 20 peers hosts or networks for IP security IPSec IKE Each of the entries in the IPSec Security Association table controls both Main Mode and Quick Mode configuration for a single peer Note You can also configure the IP Security Associations table using the ini file table parameter IPsecSATable refer to Security Parameters on page 232 gt To configure the IPSec Association table 1 Open the IP Security Associations Table page Configuration tab gt Security Settings menu gt IPSec Association Table Due to the length of the table the figure below shows sections of this table Figure 3 55 IP Security Associations Table Page Operational Authentication Source Destination Mode Remote Endpoint Addr Method Shared Key Port Port O IpSec SA Lifetime ja Dead Peer Detection Secs IpSec SA Lifetime Kbs Mode Remote Tunnel Addr Index Protoco
173. transfer to function in remote PBX extensions Hold must be disabled at the FXS device i e Enable Hold 0 and hook flash must be transferred from the FXS to the FXO HookFlashOption 4 Version 6 0 397 March 2010 7a tal AudioCodes MediaPack Series 9 4 3 6 FXO Gateway Configuration The procedure below describes how to configure the FXO interface to which the PBX is directly connected gt To configure the FXO interface 1 In the Endpoint Phone Numbers page refer to Configuring the Endpoint Phone Numbers on page 143 assign the phone numbers 200 to 207 to the device s FXO endpoints Figure 9 14 Assigning Phone Numbers to FXO Ports a Channel s Phone Number Hunt Group ID 1 8 200 2 In the Automatic Dialing page enter the phone numbers of the FXS device in the Destination Phone Number fields When a ringing signal is detected at Port 1 the FXO device automatically dials the number 100 Figure 9 15 FXO Automatic Dialing Configuration Gateway Destination Phone Auto Dial Port Number Status Port 1 FXO Enable v Port 2 FXO Enable Port 3 FXO Enable Port 4 FXO Enable Port 5 FXO Port 6 FXO Enable Port 7 FXO Enable Port FXO Enable 3 In the Tel to IP Routing page enter 10 in the Destination Phone Prefix field
174. transport modes INFO messages NOTIFY and RFC 2833 in proper payload type or as part of the audio stream To exclude RFC 2833 Telephony event parameter from the device s SDP set RxDTMFOption to 0 in the ini file The following parameters affect the way the device handles the DTMF digits m TxDTMFOption RxDTMFOption and RFC2833PayloadType m MGCPDTMFDetectionPoint DTMFVolume DTMFTransportType DTMFDigitLength and DTMFInterDigitInterval Version 6 0 385 March 2010 A e AudioCodes MediaPack Series 9 4 9 4 1 9 4 2 9 4 2 1 FXS and FXO Capabilities FXS FXO Coefficient Types The FXS Coefficients and FXO Coefficients can be defined as one of the following types m US line type of 600 ohm AC impedance and 40 V RMS ringing voltage for REN 2 m European standard TBR21 These types can be defined using the the ini file parameters FXSCountryCoefficients for FXS and CountryCoefficients for FXO or in the Web Analog Settings page refer to Configuring the Analog Settings on page 64 These Coefficient types are used to increase return loss and trans hybrid loss performance for two telephony line type interfaces US or European This adaptation is performed by modifying the telephony interface characteristics This means for example that changing impedance matching or hybrid balance doesn t require hardware modifications so that a single device is able to meet requirements for different markets The d
175. with information for configuring and operating the VoIP analog MediaPack series devices listed in the table below Table 1 1 Supported MediaPack Series Configurations Product Name FXS FXO ee pa ak MP 124 v a 7 MP 118 v F Jsa MP 114 7 v 24 P MP 112 p x F 1 1 The MP 112 differs from the MP 114 and MP 118 in that its configuration excludes the RS 232 connector Lifeline option and outdoor protection Gateway Description The MediaPack series analog Voice over IP VoIP Session Initiation Protocol SIP media gateways hereafter referred to as device are cost effective cutting edge technology products These stand alone analog VoIP devices provide superior voice technology for connecting legacy telephones fax machines and Private Branch Exchange PBX systems to IP based telephony networks as well as for integration with new IP based PBX architectures These devices are designed and tested to be fully interoperable with leading softswitches and SIP servers The device is best suited for small and medium sized enterprises SME branch offices or residential media gateway solutions The device enables users to make local or international telephone and or fax calls over the Internet between distributed company offices using their existing telephones and fax These calls are routed over the existing network ensuring that voice traffic uses minimum bandwidth The device also provides SIP trunking capabilities for Enterprises o
176. 0 ca AudioCodes Parameter Web Port EMS Destination Port Web EMS Transport Type Web Dest IP Group ID EMS Destination IP Group ID IP Profile ID Status Web EMS Charge Code SIP User s Manual MediaPack Series Description The ENUM reply includes a SIP URI used as the Reguest URI in the outgoing INVITE and for routing if a proxy is not used The IP address can include the following wildcards vy x represents single digits For example 10 8 8 xx depicts all addresses between 10 8 8 10 and 10 8 8 99 v represents any number between 0 and 255 For example 10 8 8 depicts all addresses between 10 8 8 0 and 10 8 8 255 The destination port to where you want to route the call The transport layer type used for sending the IP calls 1 Not Configured 0 UDP 1 TCP 2 TLS Note When set to Not Configured 1 the transport type defined by the parameter SIPTransportType is used The IP Group 1 9 to where you want to route the call The SIP INVITE message is sent to the IP address defined for the Proxy Set ID associated with the selected IP Group Notes If you choose an IP Group you do not need to configure a destination IP address However if both parameters are configured in this table the INVITE message is sent only to the IP Group and not the defined IP address Ifthe parameter AlwaysUseRouteTable is set to 1 refer to Configuring the IP Groups on page 91
177. 0 1 RoutingTableHopsCountColumn 20 20 Web Destination IP Address Specifies the IP address of the destination EMS Destination IP host network RoutingTableDestinationsColumn Note For this parameter to take effect a device reset is reguired Web Destination Mask Specifies the subnet mask of the destination EMS Prefix Length host network Routing TableDestinationMasksColumn Note For this parameter to take effect a device reset is required Version 6 0 211 March 2010 ca AudioCodes Parameter Web Gateway IP Address EMS Next Hop RoutingTableGatewaysColumn Web Metric EMS Primary Routing Metric RoutingTableHopsCountColumn Web Interface EMS Interface Index RoutingTablelnterfacesColumn MediaPack Series Description The IP address of the router next hop to which the packets are sent if their destination matches the rules in the adjacent columns Notes For this parameter to take effect a device reset is reguired The Gateway address must be in the same subnet as configured on the Multiple Interface Table page refer to Configuring the Multiple Interface Table on page 50 The maximum number of times a packet can be forwarded hops between the device and destination typically up to 20 Notes For this parameter to take effect a device reset is reguired This parameter must be set to a number greater than 0 for the routing rule to be valid Routing entries with
178. 0 375 March 2010 7a tall AudioCodes MediaPack Series The calling number of outgoing Tel to IP calls is first translated to an IP number and then if defined the manipulation rules are performed The Display Name is used in the From header in addition to the IP number The called number of incoming IP to Tel calls is translated to a PBX extension only after manipulation rules if defined are performed SIP User s Manual 376 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 IP Telephony Capabilities This section describes the device s main IP telephony capabilities 9 1 Dialing Plan Features This section discusses various dialing plan features supported by the device m Dialing plan notations refer to Dialing Plan Notation for Routing and Manipulation on page 377 m Digit mapping refer to Digit Mapping on page 379 External Dial Plan file containing dial plans refer to External Dial Plan File on page 380 9 1 1 Dialing Plan Notation for Routing and Manipulation The device supports flexible dialing plan notations for representing digits single or multiple entered for destination and source prefixes of phone numbers and SIP URI user names in the routing tables Table 9 1 Dialing Plan Notations Notation Description Example n m Represents a range of 5551200 5551300 represents all numbers from numbers 5551200 to 5551300 Note Range of letters 123 100 200 represents al
179. 0 433 March 2010 A tall AudioCodes MediaPack Series Table 9 4 Special Information Tones SITs Reported by the device Special Description First Tone Second Tone Third Tone Information Frequency Frequency Frequency Tones SITs Duration Duration Duration Name Hz ms Hz ms Hz ms NC1 No circuit found 985 2 380 1428 5 380 1776 7 380 Ic Operator intercept 913 8 274 1370 6 274 1776 7 380 vc Vacant circuit non 985 2 380 1370 6 274 1776 7 380 registered number RO1 Reorder system 913 8 274 1428 5 380 1776 7 380 busy NC 913 8 380 1370 6 380 1776 7 380 RO 985 2 274 1370 6 380 1776 7 380 1O 913 8 380 1428 5 274 1776 7 380 For example INFO sip 5001 10 33 2 36 SIP 2 0 Via SIP 2 0 UDP 10 33 45 65 branch z9hG4bKac2042168670 Max Forwards 70 From lt sip 5000 10 33 45 65 user phone gt tag 1c1915542705 To lt sip 5001e10 33 2 36 user phone gt tag WOJNIDDPCOKAPIDSCOTG Call ID AITFHPETLLMVVFWPDXUHD 10 33 2 36 CSeq 1 TNFO Contact lt sip 2206 10 33 45 65 gt Supported em timer replaces path resource priority Content Type application x detect Content Length 28 Type CPT SubType SIT IC The X Detect event notification process is as follows 1 For IP to Tel or Tel to IP calls the device receives a SIP request message using the X Detect header that the remote party wishes to detect events on the media stream For incoming IP to Tel calls the request must be indicated in the initial IN
180. 0 user phone SIP 2 0 Note After the cic prefix is added the IP to Hunt Group Routing Table can be used to route this call to a specific Hunt Group The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to the Tel 6 15 2 Alternative Routing Parameters The alternative routing parameters are described in the table below Table 6 55 Alternative Routing Parameters Parameter Web EMS Redundant Routing Mode RedundantRoutingMode Web Enable Alt Routing Tel to IP EMS Enable Alternative Routing AltRoutingTel2IPEnable Version 6 0 Description Determines the type of redundant routing mechanism when a call can t be completed using the main route 0 Disable No redundant routing is used If the call can t be completed using the main route using the active Proxy or the first matching rule in the Routing table the call is disconnected 1 Routing Table Internal routing table is used to locate a redundant route default 2 Proxy Proxy list is used to locate a redundant route Note To implement the Redundant Routing Mode mechansim you first need to configure the parameter AltRouteCauseTEL2IP Reasons for Alternative Routing table Enables the Alternative Routing feature for Tel to IP calls 0 Disable Disables the Alternative Routing feature default 1 Enable Enables the Alternative Routing feature 2 Status Only
181. 1 Configuration is saved to flash memory default Auxiliary and Configuration File Name Parameters Web EMS Call Progress Tones The name of the file containing the Call Progress Tones definitions File Refer to the Product Reference Manual for additional information CallProgressTonesFilename on how to create and load this file Note For this parameter to take effect a device reset is reguired Web EMS Prerecorded Tones The name and path of the file containing the Prerecorded Tones File PrerecordedTonesFileName Note For this parameter to take effect a device reset is reguired UserlnfoFileName The name and path of the file containing the User Information data Version 6 0 361 March 2010 K tal AudioCodes MediaPack Series 6 17 2 Automatic Update Parameters The automatic update of software and configuration files parameters are described in the table below Table 6 62 Automatic Update of Software and Configuration Files Parameters Parameter Description General Automatic Update Parameters AutoUpdateCmpFile Enables or disables the Automatic Update mechanism for the cmp file 0 The Automatic Update mechanism doesn t apply to the cmp file default 1 The Automatic Update mechanism includes the cmp file Note For this parameter to take effect a device reset is required AutoUpdateFrequency Determines the number of minutes the device waits between automatic updates The default value
182. 1 DH group 2 Notes Each row in the table refers to a different IKE peer To support more than one Encryption Authentication DH Group proposal for each proposal specify the relevant parameters in the Format line The proposal list must be contiguous Fora detailed description of this table and to configure the table using the Web interface refer to Configuring the IP Security Proposal Table on page 79 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 6 4 7 OCSP Parameters The Online Certificate Status Protocol OCSP parameters are described in the table below Parameter EMS OCSP Enable OCSPEnable EMS OCSP Server IP OCSPServerlP Table 6 24 OCSP Parameters Description Enables or disables certificate checking using OCSP 0 Disable default 1 Enable For a description of OCSP refer to the Product Reference Manual Defines the IP address of the OCSP server The default IP address is 0 0 0 0 OCSPSecondaryServerlP Defines the IP address in dotted decimal notation of the secondary Version 6 0 OCSP server optional The default IP address is 0 0 0 0 239 March 2010 ca AudioCodes MediaPack Series Parameter Description EMS OCSP Server Port Defines the OCSP server s TCP port number OCSPServerPort The default port number is 2560 EMS OCSP Default Determines the default OCSP behavior w
183. 1 SIP Software Package The table below lists the device s standard SIP software package File Name Table 11 1 Software Package Description Firmware RAM CMP File MP124 SIP xxx cmp MP118 SIP xxx cmp Image file containing the software for the MP 124 FXS device Common Image file Image file containing the software for MP 11x FXS devices ini Configuration Files SIPgw_MP 124 ini SIPgw_fxs_MP118 ini SIPgw_fxs_MP114 ini SIPgw_fxs_MP112 ini Usa tones xx dat Usa tones xx ini Utilities DConvert ACSyslog BootP CPTWizard MIB Files Version 6 0 Sample ini file for MP 124 FXS device Sample ini file for MP 118 FXS devices Sample ini file for MP 114 FXS devices Sample ini file for MP 112 FXS devices Default loadable Call Progress Tones dat file Call Progress Tones ini file used to create dat file TrunkPack Downloadable Conversion Utility to create Call Progress Tones files Syslog server BootP TFTP configuration utility Call Progress Tones Wizard MIB library for SNMP browser The device is supplied with a cmp file pre installed on its flash memory However if you are an AudioCodes registered customer you can obtain the latest cmp version files as well as documentation and other software such as the ini and MIB files and Utilities from AudioCodes Web site at www audiocodes com support customer registration is performed online at this Web site If you are not a direct customer of AudioCodes
184. 10 33 37 78 CSeg 1 REGISTER Contact lt sip ContactUsere10 33 37 78 gt expires 3600 Expires 3600 User Agent Sip Gateway v 6 00A 008 002 Content Length 0 Notes The Hunt Group account registration is not affected by the parameter IsRegisterNeeded f registration to an IP Group s fails for all the accounts defined in this table for a specific Hunt Group and if this Hunt Group includes all the channels in the Hunt Group the Hunt Group is set to Out Of Service if the parameter OOSOnRegistrationFail is set to 1 refer to Proxy 8 Registration Parameters on page 96 Defines the AOR user name It appears in REGISTER From To headers as ContactUser HostName and in INVITE 200 OK Contact headers as ContactUser lt device s IP address gt If not configured the Contact User parameter from the IP Group Table page is used instead Note If registration fails then the user part in the INVITE Contact header contains the source party number Note This parameter is not applicable 95 March 2010 ca AudioCodes 3 3 4 4 3 Configuring Proxy and Registration Parameters MediaPack Series The Proxy 8 Registration page allows you to configure parameters that are associated with Proxy and Registration For a description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 Note To view whether the device or its endpoints have registered to a SIP Registrar Proxy server r
185. 107 March 2010 ca AudioCodes gt To configure the IP Profile settings MediaPack Series 1 Open the IP Profile Settings page Configuration tab gt Protocol Configuration menu gt Coders And Profile Definitions submenu gt IP Profile Settings Figure 3 67 IP Profile Settings Page v Profile ID Profile Name v Common Parameters RTP IP DiffServ Signaling DiffServ Disconnect on Broken Connection Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Echo Canceler Input Gain 32 to 31 dBj Voice Volume 32 to 31 dB Dynamic Jitter Buffer Minimum Delay msec Enable 0 0 w Gateway Parameters Fax Signaling Method Play Ringback Tone to IP Enable Early Media Media Security Behavior CNG Detector Mode Modems Transport Type NSE Mode Number of Calls Limit Progress Indicator to IP Profile Preference Coder Group Remote RTP Base UDP Port First Tx DTMF Option Second Tx DTMF Option Declare RFC 2833 in SDP Enable Hold Copy Destination Number to Redirect Number No Fax Don t Play Disable Disable Preferable Disable Enable Bypass Disable SS SSNS NS NSS 1 Not Configured 1 Default Coder Group 0 Mot Supported Mot Supported Yes Enab
186. 13 SIP User s Manual 6 Configuration Parameters Reference Parameter EMS Polarity Reversal Type PolarityReversalType EMS Current Disconnect Duration CurrentDisconnectDuration CurrentDisconnectDefaultThreshold TimeToSampleAnalogLineVoltage Version 6 0 Description disconnect threshold FXO only is determined by the parameter CurrentDisconnectDefaultThreshold The freguency at which the analog line voltage is sampled is determined by the parameter TimeToSampleAnalogLineVoltage Defines the voltage change slope during polarity reversal or wink 0 Soft reverse polarity default 1 Hard reverse polarity Notes This parameter is applicable only to FXS interfaces Some Caller ID signals use reversal polarity and or Wink signals In these cases it is recommended to set the parameter PolarityReversalType to 1 Hard For this parameter to take effect a device reset is required The duration in msec of the current disconnect pulse The range is 200 to 1500 The default is 900 Notes This parameter is applicable for FXS and FXO interfaces The FXO interface detection window is 100 msec below the parameter s value and 350 msec above the parameter s value For example if this parameter is set to 400 msec then the detection window is 300 to 750 msec For this parameter to take effect a device reset is required Determines the line voltage threshold at which a curren
187. 16 255 255 0 0 192 168 0 1 0 1 16 255 255 0 0 192 168 0 2 0 1 16 255 255 0 0 192 168 0 3 0 1 16 255 255 0 0 192 168 0 25 0 1 10 8 2 2 Routing Table Columns Each row of the Routing table defines a routing rule Traffic destined to the subnet specified in the routing rule is re directed to a specified gateway reachable through a specified interface 10 8 2 2 1Destination Column This column defines the destination of the route rule The destination can be a single host or a whole subnet depending on the Prefix Length Subnet Mask specified for this routing rule Version 6 0 457 March 2010 Aa L l AudioCodes MediaPack Series 10 8 2 2 2Prefix Length and Subnet Mask Columns These two columns offer two notations for the mask You can either enable the Subnet Mask in dotted decimal notation or the CIDR style representation Please note that only one of these is needed If both are specified the Prefix Length column overrides the Subnet Mask column Figure 10 3 Prefix Length and Subnet Masks Columns Tani j Pe aa i kui e i n 201 201 85 14 164 255 255 255 252 192168025 0 1 P m Even though the Subnet Mask column indicates a subnet mask of 255 255 255 252 the actual mask will be 255 255 0 0 as the Prefix Length column overrides the Subnet Mask column 10 8 2 2 3Gateway Column The Gateway column defines the IP Address of the next hop used for traffic destined to the subnet as specified
188. 2 m BellModemTransportType 2 Fax Modem Transparent with Events Mode In this mode fax and modem signals are transferred using the current voice coder with the following automatic adaptations m Echo Canceller on or off for modems m Echo Canceller Non Linear Processor Mode off m Jitter buffering optimizations To configure fax modem transparent with events mode perform the following configurations IsFaxUsed 0 FaxTransportMode 3 V21ModemTransportType 3 V22ModemTransportType 3 V23ModemTransportType 3 V32ModemTransportType 3 V34ModemTransportType 3 BellModemTransportType 3 Fax Modem Transparent Mode In this mode fax and modem signals are transferred using the current voice coder without notifications to the user and without automatic adaptations It s possible to use the Profiles mechanism refer to Coders and Profile Definitions on page 101 to apply certain adaptations to the channel used for fax modem e g to use the coder G 711 to set the jitter buffer optimization factor to 13 and to enable echo cancellation for fax and disable it for modem To configure fax modem transparent mode use the following parameters IsFaxUsed 0 FaxTransportMode 0 V21ModemTransportType 0 V22ModemTransportType 0 V23ModemTransportType 0 V32ModemTransportType 0 V34ModemTransportType 0 BellModemTransportType 0 Additional configuration parameters e CodersGroup Version 6 0 405
189. 20 seconds Note For this parameter to take effect a device reset is required Silence detection method 0 None Silence detection option is disabled 1 Packets Count According to packet count 2 Voice Energy Detectors According to energy and voice detectors default 3 All According to packet count and energy and voice detectors Note For this parameter to take effect a device reset is required 315 March 2010 A K tal AudioCodes MediaPack Series Parameter Description FarEndDisconnectSilenceThreshold Threshold of the packet count in percentages below which is considered silence by the device The valid range is 1 to 100 The default is 8 Notes This parameter is applicable only if silence is detected according to packet count FarEndDisconnectSilenceMethod is set to 1 For this parameter to take effect a device reset is required BrokenConnectionDuringSilence Enables the generation of the BrokenConnection event during a silence period if the channel s NoOp feature is enabled using the parameter NoOpEnable and if the channel stops receiving NoOp RTP packets 0 Disable default 1 Enable Web Disconnect Call on Busy Tone Determines whether a call is disconnected upon detection Detection of a busy tone She goa On Detection End 0 Disable Do not disconnect call on detection of busy tone 1 Enable Call is released if busy or reorder fast
190. 201 10 8 1 You can also specify the selected port in the format lt IP address gt lt port gt If you enable Proxy Redundancy by setting the parameter EnableProxyKeepAlive to 1 or 2 the device can operate with multiple Proxy servers If there is no response from the first primary Proxy defined in the list the device attempts to communicate with the other redundant Proxies in the list When a redundant Proxy is located the device either continues operating with it until the next failure occurs or reverts to the primary Proxy refer to the parameter ProxyRedundancyMode If none of the Proxy servers respond the device goes over the list again The device also provides real time switching Hot Swap mode between the primary and redundant proxies refer to the parameter IsProxyHotSwap If the first Proxy doesn t respond to the INVITE message the same INVITE message is immediately sent to the next Proxy in the list The same logic applies to REGISTER messages if RegistrarlP is not defined Notes f EnableProxyKeepAlive is set to 1 or 2 the device monitors the connection with the Proxies by using keep alive messages OPTIONS or REGISTER To use Proxy Redundancy you must specify one or more redundant Proxies When a port number is specified e g domain com 5080 DNS NAPTR SRV queries aren t performed even if ProxyDNSQueryType is set to 1 or 2 Transport Type The transport type per Proxy server Proxylp_Transport
191. 2010 A c tal AudioCodes MediaPack Series Parameter Description requires prior configuration of the server certificate and root CA refer to Configuring the Certificates on page 73 The parameter 802 1xUsername is used to identify the device however 802 1xPassword is ignored Note The configured mode must match the configuration of the Access server e g RADIUS server Web 802 1x Username Username for IEEE 802 1x support EMS User Name The valid value is a string of up to 32 characters The default is an 802 1xUsername empty string Web 802 1x Password Password for IEEE 802 1x support EMS Password The valid value is a string of up to 32 characters The default is an 802 1xPassword empty string Web 802 1x Verify Peer Verify Peer Certificate for IEEE 802 1x support Certificate EMS Verify Peer Certificate 0 Disable default 802 1xVerifyPeerCertificate 1 Enable 6 1 2 Multiple IP Interfaces and VLAN Parameters The IP network interfaces and VLAN parameters are described in the table below Table 6 2 IP Network Interfaces and VLAN Parameters Parameter Description Web Multiple Interface Table EMS IP Interface Settings InterfaceTable This ini file table parameter configures the Multiple Interface table for configuring logical IP addresses The format of this parameter is as follows InterfaceTable FORMAT InterfaceTable_Index InterfaceTable_ApplicationTypes InterfaceTable_InterfaceMod
192. 24 inbound call routing rules The device uses these rules for routing incoming IP calls to Hunt Groups The specific channel pertaining to the Hunt Group to which the call is routed is determined according to the Hunt Group s channel selection mode The channel selection mode can be defined per Hunt Group refer to Configuring Hunt Group Settings on page 85 or for allHunt Groups using the global parameter ChannelSelectMode This table provides two main areas for defining a routing rule m Matching Characteristics user defined characteristics of the incoming IP call are defined in this area If the characteristics match a table entry the rule is used to route the call One or more characteristics can be defined for the rule such as source calling destination called telephone number prefix and source IP address from where call received m Destination user defined destination If the call matches the characteristics the device routes the call to this destination The destination is a selected Hunt Group When a call release reason defined in Configuring Reasons for Alternative Routing on page 124 is received for a specific IP to Tel call an alternative Hunt Group for that call can be configured This is done by configuring an additional routing rule for the same call characteristics but with a different Hunt Group ID You can also configure the IP to Hunt Group Routing Table using the ini file table parameter PSTNPrefi
193. 4 EN61000 3 3 EN61000 3 2 VCCI Class X1 equals to class B EN60950 1 Safety of information technology equipment UL60950 1 Including compliance to section 6 over voltage protection TRR 21 TIA 968 469 March 2010 7a u wi AudioCodes CPE 8 Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 6 0 C A AudioCodes www audiocodes com
194. 50Cause parameter is used This parameter is applicable only to FXO interfaces Determines the A 850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT VC Vacant Circuit non registered number Special Information Tone is detected from the Tel for IP to Tel calls The valid range is 0 to 127 The default value is 1 not configured Notes When not configured i e default the SITQ850Cause parameter is used This parameter is applicable only to FXO interfaces Determines the Q 850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT RO Reorder System Busy Special Information Tone is detected from the Tel for IP to Tel calls The valid range is 0 to 127 The default value is 1 not configured Notes When not configured i e default the SITQ850Cause parameter is used This parameter is applicable only to FXO interfaces Out of Service Busy Out Parameters Web EMS Enable Busy Out EnableBusyOut Version 6 0 Determines whether the Busy Out feature is enabled 0 Disable Busy out feature is not used default 1 Enable Busy ou feature is enabled When Busy Out is enabled and certain scenarios exist the device performs the following A reorder tone configured by the parameter FXSOOSBehavior is played when the phone is off hooked These behaviors are performed upon one of the following scenario
195. 5413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description 15 BootP retries indefinitely BootPSelectiveEnable Enables the Selective BootP mechanism 1 Enabled 0 Disabled default The Selective BootP mechanism available from Boot version 1 92 enables the device s integral BootP client to filter unsolicited BootP DHCP replies accepts only BootP replies that contain the text AUDC in the vendor specific information field This option is useful in environments where enterprise BootP DHCP servers provide undesired responses to the device s BootP reguests Notes For this parameter to take effect a device reset is required When working with DHCP i e the parameter DHCPEnable is set to 1 the selective BootP feature must be disabled BootPDelay The interval between the device s startup and the first BootP DHCP request that is issued by the device 1 1 second default 2 3 second 3 6 second 4 30 second 5 60 second Note For this parameter to take effect a device reset is required ExtBootPReqEnable 0 Disable default 1 Enable extended information to be sent in BootP request If enabled the device uses the Vendor Specific Information field in the BootP request to provide device related initial startup information such as blade type current IP address software version For a full list of the Vendor Specific Informa
196. 55skKw 6000 2000 10 8 201 108 CSeg 18153 INVITE Contact lt sip 8000 10 8 201 108 user phone gt User Agent Audiocodes Sip Gateway MediaPack v 6 00 010 006 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 208 v 0 o AudiocodesGW 18132 74003 IN IP4 10 8 201 108 s Phone Call c IN IP4 10 8 201 108 t 0 0 Version 6 0 421 March 2010 AM tal AudioCodes MediaPack Series m audio 4000 RTP AVP 8 96 a rtpmap 8 pcma 8000 a rtpmap 96 telephone event 8000 E BMEDSS W515 a ptime 20 m F2 TRYING 10 8 201 161 gt gt 10 8 201 108 SIP 2 0 100 Trying Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 6000 10 8 201 108 gt tag 1c5354 Woe Sip 2000 10 8 20 161s Call ID 534366556655skKw 6000 2000 10 8 201 108 Server Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeq 18153 INVITE Content Length 0 m F3 RINGING 180 10 8 201 161 gt gt 10 8 201 108 SIP 2 0 180 Ringing Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 6000 10 8 201 108 gt tag 1c5354 To lt sip 2000e10 8 201 161 gt tag 1c7345 Call ID 534366556655skKw 6000 2000 10 8 201 108 Server Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeg 18153 INVITE Supported 100rel em Content Length 0 Note Phone 2000 answers the call and then sends a 200 OK message to device
197. 7a u ei AudioCodes CPE 8 Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 6 0 Document LTRT 65413 March 2010 1 2 3 SIP User s Manual Contents Table of Contents i 12 2 1 2 MP 124 o Feature SIP OVEM PA E a rere L 1 3 Configuration Concepts sssssssssssssssssssssessnessnesseesnesseeseeeserseersnersnes 21 Web Based Management csccssccsscsscssesseesseesseesseesseesseeeseessesesneeeneeeneeeaee 23 3 1 ao Acquainted with the Web Interface Version 6 0 3 March 2010 7a e AudioCodes MediaPack Series 3 3 2 5 Configuring the Analog Settings ccccccsscsesssssecsssccsscenssesessseessses OF 3 3 2 6 Configuring Media Security 3 3 3 Security Settings i 3391 Configuring the Web User Accounts 3 3 3 2 Configuring the Web and Telnet Acces 3 3 3 3 Configuring the Firewall Settings ass 3 3 3 4 Configuring the Certificates eaaeo 3 3 3 5 Configuring the 802 1x Settings ANDI 3 3 3 6 Configuring the General Security Se ngs 3 3 3 7 Configuring the IP Security Proposal Table 3 3 3 8 Configuring the IP pm Associations TANG s sa neadekuaasane Oe 3 3 4 Protocol Configuration san RIN ata asa Ska sounds E 3 3 4 1 Enabling Applications 3 3 4 2 Hunt Group re 3 3 4 3 Protocol Definition TEP PPR PS PRO PR PRE 3 3 4 4 Proxies Registration IP Groups EEOAE EE EE AE u 3 3 4 5
198. 8 0 and 10 8 8 255 118 Document LTRT 65413 SIP User s Manual Parameter Web Stripped Digits From Left EMS Number Of Stripped Digits Web Stripped Digits From Right EMS Number Of Stripped Digits Web Prefix to Add EMS Prefix Suffix To Add Web Suffix to Add EMS Prefix Suffix To Add Web EMS Number of Digits to Leave Web Presentation EMS Is Presentation Restricted Version 6 0 3 Web Based Management Description Number of digits to remove from the left of the telephone number prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 1234 Number of digits to remove from the right of the telephone number prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 5551 The number or string that you want added to the front of the telephone number For example if you enter 9 and the phone number is 1234 the new number is 91234 The number or string that you want added to the end of the telephone number For example if you enter 00 and the phone number is 1234 the new number is 123400 The number of digits that you want to retain from the right of the phone number Determines whether Caller ID is permitted Not Configured privacy is determined according to the Caller ID table refer to Configuring Caller Display Information on page 138 Allowed sends Caller ID information when a call is made using these dest
199. 8 mode using SIP Re INVITE messages set IsFaxUsed to 1 Additional configuration parameters include the following E FaxRelayEnhancedRedundancyDepth mM FaxRelayRedundancyDepth m FaxRelayECMEnable m FaxRelayMaxRate The terminating gateway sends T 38 packets immediately after the T 38 capabilities are negotiated in SIP However the originating device by default sends T 38 assuming the T 38 capabilities are negotiated in SIP only after it receives T 38 packets from the remote device This default behavior cannot be used when the originating device is located behind a firewall that blocks incoming T 38 packets on ports that have not yet received T 38 packets from the internal network To resolve this problem the device should be configured to send CNG packets in T 38 upon CNG signal detection CNGDetectorMode 1 Version 6 0 401 March 2010 7a E tall AudioCodes MediaPack Series 9 6 2 1 2 Automatically Switching to T 38 Mode without SIP Re INVITE 9 6 2 2 9 6 2 3 In the Automatically Switching to T 38 Mode without SIP Re INVITE mode when a fax signal is detected the channel automatically switches from the current voice coder to answer tone mode and then to T 38 compliant fax relay mode To configure automatic T 38 mode perform the following configurations m IsFaxUsed 0 E FaxTransportMode 1 m Additional configuration parameters e FaxRelayEnhancedRedundancyDepth e FaxRelayRedundancyDepth e FaxRelayECMEnable e
200. 9 5 Configuring Caller ID Permissions The Caller ID Permissions page allows you to enable or disable per port the Caller ID generation for FXS interfaces and detection for FXO interfaces If a port isn t configured its Caller ID generation detection are determined according to the global parameter EnableCallerID described in Configuring Supplementary Services on page 111 Note You can also configure the Caller ID Permissions table using the ini file table parameter EnableCallerlD gt To configure Caller ID Permissions per port 1 Open the Caller ID Permissions page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Caller ID Permissions page item Figure 3 88 Caller ID Permissions Page Gateway Port FXS FXS FXS FXS FXO FXO FXO FXO 2 From the Caller ID drop down list select one of the following e Enable Enables Caller ID generation FXS or detection FXO for the specific port e Disable Caller ID generation FXS or detection FXO for the specific port is disabled e Not defined Caller ID generation FXS or detection FXO for the specific port is determined according to the parameter Enable Caller ID described in Configuring Supplementary Services on page 111 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 141 Ma
201. AT mechanism must be enabled for this parameter to take effect i e the parameter DisableNAT is set to 0 For information on RTP Multiplexing refer to RTP Multiplexing ThroughPacket on page 440 EnableUDPPortTranslation 0 Disable UDP port translation default 1 Enable UDP port translation When enabled the device compares the source UDP port of the first incoming packet to the remote UDP port stated in the opening of the channel If the two UDP ports don t match the NAT mechanism is activated Consequently the remote UDP port of the outgoing stream is replaced by the source UDP port of the first incoming packet Notes For this parameter to take effect a device reset is required The NAT mechanism and the IP address translation must be enabled for this parameter to take effect i e set the parameter DisableNAT to 0 and the parameter EnablelpAddrTranslation to 1 SIP User s Manual 216 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 1 6 NFS Parameters The Network File Systems NFS configuration parameters are described in the table below Parameter NFSBasePort Web NFS Table EMS NFS Settings NFSServers Version 6 0 Table 6 6 NFS Parameters Description Start of the range of numbers used for local UDP ports used by the NFS client The maximum number of local ports is maximum channels plus maximum NFS servers The valid range is 0 to
202. Account Attribute User Name Password Access Level Case Sensitive Case Sensitive Primary Account Admin Admin Security Administrator Note The Access Level cannot be changed for this account type Secondary Account User User User Monitor gt To change the Web user accounts attributes 1 Open the Web User Accounts page Configuration tab gt Security Settings menu gt Web User Accounts page item Figure 3 46 WEB User Accounts Page for Users with Security Administrator Privileges Current Logged User Admin w Account Data for User Admin User Name Change User Name Access Level wv Fill in the following 3 fields to change the password Current Password New Password Confirm New Password Change Password Account Data for User User 2 User Name User 2 Change User Name Access Level Administrator Change Access Level w Fill in the following 3 fields to change the password Current Password New Password Confirm New Password Change Password Note If you are logged into the Web interface as the Security Administrator both Web user accounts are displayed on the Web User Accounts page as shown above If you are logged in with the secondary user account only the details of the secondary account are displayed on the page Version 6 0 67 March 2010 Aa c tal AudioCodes MediaPack Series 2 To change the acce
203. AudioCodes MediaPack Series 3 3 1 4 Configuring the IP Routing Table The IP Routing Table page allows you to define up to 50 static IP routing rules for the device For example you can define static routing rules for the OAMP and Control networks since a default gateway is supported only for the Media traffic network Before sending an IP packet the device searches this table for an entry that matches the requested destination host network If such an entry is found the device sends the packet to the indicated router If no explicit entry is found the packet is sent to the default gateway refer to Configuring the Multiple Interface Table on page 50 gt To configure static IP routing 1 Open the IP Routing Table page Configuration tab gt Network Settings menu gt IP Routing Table page item Figure 3 38 IP Routing Table Page O Delete Raw Destination IP Address Destination Mask Gateway IP Address Metric Interface Delete Selected Entries PEE ER Destination IP Address Destination Mask Gateway IP Address Metric Interface Ic M 1 Add New Entry 2 In the Add a new table entry group add a new static routing rule according to the parameters described in the table below 3 Click Add New Entry the new routing rule is added to the IP routing table To delete a routing rule from the table
204. C2833PayloadType default 1 INFO Nortel Sends DTMF digits according to IETF lt draft choudhuri sip info digit 00 gt 2 NOTIFY Sends DTMF digits according to IETF lt draft mahy sipping signaled digits 01 gt 3 INFO Cisco Sends DTMF digits according to Cisco format 4 RFC 2833 5 INFO Korea Sends DTMF digits according to Korea Telecom format 280 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Notes DTMF negotiation methods are prioritized according to the order of their appearance When out of band DTMF transfer is used 1 2 3 or 5 the parameter DTMFTransportType is automatically set to 0 DTMF digits are erased from the RTP stream When RFC 2833 4 is selected the device a Negotiates RFC 2833 payload type using local and remote SDPs b Sends DTMF packets using RFC 2833 payload type according to the payload type in the received SDP c Expects to receive RFC 2833 packets with the same payload type as configured by the parameter RFC2833PayloadType d Sends DTMF digits in transparent mode as part of the voice stream When TxDTMFOption is set to 0 the RFC 2833 payload type is set according to the parameter RFC2833PayloadType for both transmit and receive The ini file table parameter TxDTMFOption can be repeated twice for configuring the DTMF transmit methods Web EMS Tx DTMF Option
205. CM or ADPCM 3 Way conference with local mixing DiffServ TOS 802 1 P Q VLAN tagging RTP RTCP per IETF RFC 3550 and 3551 PPPoE Multiplexing aggregated RTP streams of several channels for saving network bandwith MP 112 FXS Loop start MP 114 8 MP 118 FXS FXO Loop start MP 124 FXS Loop start 467 March 2010 ca AudioCodes Function In band Signaling Out of Band Signaling Control Provisioning Protocols Security Media Control Management Physical Power Environmental Dimensions Mounting Additional Features Message Waiting Indication High Availability Ring voltage Ring Freguency Maximum Ringer Load Loop Impedance including phone impedance SIP User s Manual MediaPack Series Specification DTMF TIA 464B User defined and call progress tones DTMF Relay RFC 2833 DTMF via SIP INFO SIP RFC 3261 BootP DHCP TFTP and HTTP for Automatic Installation DHCP options 66 67 in auto update mode Remote management using Web browser EMS Element Management System SNMP V3 Syslog support RS 232 for basic configuration via CLI Voice Menu using touch tone phone for basic configuration SRTP H 235 IPSec TLS SIPS HTTPS Access List IPSec 100 240 V AC 50 60 Hz or 48V DC Note 48V DC is supported only on the MP 124D Operational 5 to 40 C 41 to 104 F Storage 25 to 85 C 13 to 185 F Humidity 10 to 90 non condensing MP 112 42 x 172 x 220 mm
206. Caller ID Parameters 289 6 8 2 Call Waiting Parameters 294 6 8 3 Call Forwarding Parameters 6 8 4 Message Waiting Indication Parameters 299 6 8 5 Call Hold Parameters 6 8 6 Call Transfer Para e 6 8 7 Three Way Conferencing Paramet 6 8 8 Emergency Call Parameters 6 8 9 FXS Call Cut Through Parame 6 8 10 Automatic Dialing Parameters 6 8 11 Direct Inward Dialing Parameters 307 Pe MLPP Maj sb vn ne PE EEE EEG BOB 6 9 6 10 6 11 1 Telephony Tone Parameters 6 11 2 Tone Detection Parameters 6 11 3 Metering Tone Parameters AS PTEM ROVNY SAOP en Poe Ve oVe my Mne 6 12 Telephone Keypad Seguence Parameters ZOO EV V R EROT PO KP 324 6 13 General FXO ParameterS ccccccccceceeceecececeeecececeeccseceeceussescasesesessaesesessseeersO2O 6 14 FXS Parameters 6 15 Hunt Groups Number Kanda and d Routing F Parameters EEE 331 6 15 1 Hunt Groups and Routing PAKAM ler inaina niidina 331 6 15 2 Alternative Routing Parameters is 6 15 3 Number Manipulation Parameters c cccccsccsssccssccssecsseceecesscesseeteestseseteeseee O 41 6 16 Channel Paramotels sorciernereeree narr aene OF EO 6 16 1 Voice Parameters 6 16 2 Fax and Modem Parameters 6 16 3 DTMF Parameters 6 16 4 RTP RTCP and T 38 Parameters 6 17 Auxiliary and Configuration Files Parameters eeeeeeeeeeeeeeeee een GO 6 17 1 Auxiliary Configuration File
207. Conference ID Three Way Conference Mode l Max 3 Way Conference on Board Calls Non Allocatable Ports Disable conf AudioCodes Media Server 2 0 MLPP Call Priority Mode MLPP Diffserv Precedence Ringing Type Disable 50 1 111 March 2010 7a c tal AudioCodes MediaPack Series 2 Configure the parameters as required 3 Click the Submit button to save your changes or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe unsubscribe to the MWI server 4 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 6 3 Configuring Metering Tones The FXS interfaces can generate 12 16 KHz metering pulses towards the Tel side e g for connection to a payphone or private meter Tariff pulse rate is determined according to an internal table This capability enables users to define different tariffs according to the source destination numbers and the time of day The tariff rate includes the time interval between the generated pulses and the number of pulses generated on answer The Metering Tones page is available only for FXS interfaces Charge Code rules can be assigned to routing rules in the Tel to IP Routing refer to Configuring Tel to IP Routing on page 126 When a new call is established the Tel to IP Routing is searched for the d
208. DefaultGatewaylP The valid range is 1 to 5 The default value is 1 i e no redundant Proxy Set Enables the device to change the SIP Contact header so that it points to the SAS host and therefore the top most SIP Via header and the Contact header point to the same host 0 default Disable when relaying requests the SAS agent adds a new Via header with the SAS IP address as the top most Via header and retains the original Contact header Thus the top most Via header and the Contact header point to different hosts 1 Enable the device changes the Contact header so that it points to the SAS host and therefore the top most Via header and the Contact header point to the same host Note Operating in this mode causes all incoming dialog reguests to traverse the SAS which may cause load problems Determines the Survivability mode used by the SAS application 0 Standard All incoming INVITE and REGISTER requests are forwarded to the defined Proxy list of SASProxySet in Normal mode and handled by the SAS application in Emergency mode default 1 Always Emergency The SAS application does not use Keep Alive messages towards the SASProxySet instead it always operates in Emergency mode as if no Proxy in the SASProxySet is available 311 March 2010 ca AudioCodes Parameter Web SAS Binding Mode EMS Binding Mode SASBindingMode Web SAS Emergency Numbers SASEmergencyNumbers SASEmerg
209. Description Determines whether the port number is added as a prefix to the called number for Tel to IP calls 0 No port number not added as prefix default 1 Yes port number added as prefix If enabled the port number single digit in the range 1 to 8for 8 port devices two digits in the range 01 to 24 for MP 124 is added as a prefix to the called destination phone number This option can be used to define various routing rules Determines whether the device adds the Hunt Group ID from where the call originated as the prefix to the calling number i e source number 0 No default 1 Yes Determines whether the device removes the prefix from the destination number for IP to Tel calls 0 No Don t remove prefix default 1 Yes Remove the prefix defined in the IP to Hunt Group Routing Table refer to Configuring the IP to Hunt Group Routing Table on page 131 froma telephone number for an IP to Tel call before forwarding it to Tel For example To route an incoming IP to Tel call with destination number 21100 the IP to Hunt Group Routing Table is scanned for a matching prefix If such a prefix is found e g 21 then before the call is routed to the corresponding Hunt Group the prefix 21 is removed from the original number and therefore only 100 remains Notes This parameter is applicable only if number manipulation is performed after call routing for IP to Tel ca
210. For example SRVZ2IP 0 SrvDomain 0 Dnsname1 1 1 500 Dnsname2 2 2 501 0 0 0 Notes This parameter can include up to 10 indices If the Internal SRV table is used the device first attempts to resolve a domain name using this table If the domain name isn t located the device performs an SRV resolution using an external DNS server To configure the Internal SRV table using the Web interface and for a description of the parameters in this ini file table parameter refer to Configuring the Internal SRV Table on page 134 For an explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 219 March 2010 Aa K tal AudioCodes MediaPack Series 6 1 8 DHCP Parameters The Dynamic Host Control Protocol DHCP parameters are described in the table below Table 6 8 DHCP Parameters Parameter Description Web Enable DHCP Determines whether Dynamic Host Control Protocol DHCP is EMS DHCP Enable enabled DHCPEnable 0 Disable Disable DHCP support on the device default 1 Enable Enable DHCP support on the device After the device powers up it attempts to communicate with a BootP server If a BootP server does not respond and DHCP is enabled then the device attempts to obtain its IP address and other networking parameters from the DHCP server Notes For this parameter to take effect a device reset is required After you enable the DHCP server p
211. Group The device forwards calls using this table only if no alternative IP to Tel routing has been configured or alternative routing fails and the following reason included in the SIP Diversion header of 3xx messages exists m unavailable e All FXS FXO lines pertaining to a Hunt Group are busy or unavailable Note You can also configure the Forward on Busy Trunk Destination table using the ini file parameter table ForwardOnBusyTrunkDest Version 6 0 135 March 2010 A c tal AudioCodes MediaPack Series To configure the Forward on Busy Trunk Destination table 1 Open the Forward on Busy Trunk Destination page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Forward on Busy Trunk Dest page item Figure 3 83 Forward on Busy Trunk Destination Page Index Trunk Group ID Forward Destination The figure above includes a configuration entry that forwards IP to Tel calls destined for Hunt Group ID 2 to destination IP address 10 13 5 67 Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Saving Configuration on page 161 3 3 4 9 Endpoint Settings The Endpoint Settings submenu allows you to configure analog FXS FXO port specific parameters This submenu includes the following page items Authentication refer to Configuring Authentication on page 136 Automatic Dialing refer to Configuring A
212. Group The Hunt Group submenu allows you to configure groups of channels called Hunt Groups This submenu includes the Hunt Group Settings page item refer to Configuring Configuring Hunt Group Settings on page 85 3 3 4 2 1 Configuring Hunt Group Settings The Hunt Group Settings page allows you to configure the settings of up to 24 Hunt Groups These Hunt Groups are configured in the Endpoint Phone Number Table page refer to Configuring the Endpoint Phone Numbers on page 143 This page allows you to select the method for which IP to Tel calls are assigned to channels within each Hunt Group If no method is selected for a specific Hunt Group the setting of the global parameter ChannelSelectMode takes effect In addition this page defines the method for registering Hunt Groups to selected Serving IP Group IDs if defined Note You can also configure the Hunt Group Settings table using the ini file table parameter TrunkGroupSettings refer to Number Manipulation and Routing Parameters on page 331 To configure the Hunt Group Settings table 1 Open the Hunt Group Settings page Configuration tab gt Protocol Configuration menu gt Hunt Group submenu gt Hunt Group Settings page item Figure 3 57 Hunt Group Settings Page Hunt Group ID Serving IP Channel Select Mode Registration Mode Group ID Gateway Name Contact User 1 Cyclic Ascending v Per Gateway v v v v 2 From the
213. I for outgoing SIP OPTIONS requests If no value is configured the endpoint number is used A special value is empty indicating that no user part in the Request URI host part only is used The valid range is a 30 character string The default value is an empty string Determines the SIP signaling method for establishing and transmitting a fax session after a fax is detected 0 No Fax No fax negotiation using SIP signaling Fax transport method is according to the parameter FaxTransportMode default 1 T 38 Relay Initiates T 38 fax relay 2 G 711 Transport Initiates fax modem using the coder G 711 A law Mu law with adaptations refer to Note below 246 Document LTRT 65413 SIP User s Manual Parameter Web SIP Transport Type EMS Transport Type SIPTransportType Web SIP UDP Local Port EMS Local SIP Port LocalSIPPort Web SIP TCP Local Port EMS TCP Local SIP Port TCPLocalSIPPort Web SIP TLS Local Port EMS TLS Local SIP Port TLSLocalSIPPort Version 6 0 6 Configuration Parameters Reference Description 3 Fax Fallback Initiates T 38 fax relay If the T 38 negotiation fails the device re initiates a fax session using the coder G 711 A law u law with adaptations refer to the Note below Notes Fax adaptations for options 2 and 3 Echo Canceller On Silence Compression Off Echo Canceller Non Linear Processor Mode Off Dynamic Jitter Buffer M
214. IP Configuration Parameters on page 245 ProxylP used for creating a Proxy Set ID defined with IP addresses ProxySet used for defining various attributes for the Proxy Set ID Proxy Sets can be assigned only to SERVER type IP Groups Version 6 0 97 March 2010 7a E tal AudioCodes MediaPack Series 2 de ee To add Proxy servers and configure Proxy parameters Open the Proxy Sets Table page Configuration tab gt Protocol Configuration menu gt Proxies Registration IP Groups submenu gt Proxy Sets Table page item Figure 3 63 Proxy Sets Table Page poos Proxy Set ID 0 v Proxy Address Transport Type v v d Enable Proxy Keep Alive Disable Proxy Keep Alive Time 60 Proxy Load Balancing Method Disable Is Proxy Hot Swap No SRD Index 0 From the Proxy Set ID drop down list select an ID for the desired group Configure the Proxy parameters according to the following table Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 16 Proxy Sets Table Parameters Parameter Description Web Proxy Set ID The Proxy Set identification number EMS Index The valid range is 0 to 9 i e up to ten Proxy Set ID s can be ProxySet Index configured The Proxy Set ID 0 is used as the default Proxy Set
215. IP Routing Table Description Web User Accounts Access Levels and Privileges 3 8 Default Attributes for the Web User Accounts le 3 9 Internal Firewall Parameters le 3 10 IP Security Proposals Table Configuration Parameters 3 11 Default IPSec IKE Proposals P S Ony 2 IP Security Associations Table Configur i 3 13 Hunt Group Settings Parameters 14 IP Group Parameters Table 3 15 Account Table Parameters Descrip Table 3 16 Proxy Sets Table Parameters Table 3 17 Description of Parameter Unigue to IP Profile Table 3 18 Number Manipulation Parameters Description Table 3 19 Redirect Number Tel to IP Parameters Description save V2 Table 3 20 Phone Context Parameters Description 23 Table 3 21 Tel to IP Routing Table Parameters cccccccescccsseesceeeseeeeeceeeaeseeseaeeeesetsessassatestentees 129 Table 3 Pa6 Tel Reming Table EGSGnPIOT su zd sdkkkus k k slad ban s nis aadis OA Table 3 23 Call Forward Table Table 3 24 Endpoint Phone Number Table Parameters Table 3 25 SAS Routing Table Parameters Table 3 26 SNMP Trap Destinations Parameters Table 3 27 SNMP Community Strings Parameters Table 3 28 SNMP V3 Users Param Table 3 29 Auxiliary Files Descriptions 2 Table 3 30 Ethernet Port Information Parameters Table 3 31 IP Interface Status Page Table 3 32 Device Information Page Table 3 33 Call Counters Description Table 3 34 SAS Registere Table 3 35 Table 3
216. IP address unique VLAN ID if enabled interface name and application type permitted on the interface m Control m Media m Operations Administration Maintenance and Provisioning OAMP This page also provides VLAN related parameters for enabling VLANs and for defining the Native VLAN ID VLAN ID to which incoming untagged packets are assigned For assigning VLAN priorities and Differentiated Services DiffServ for the supported Class of Service CoS refer to Configuring the QoS Settings on page 60 SIP User s Manual 50 Document LTRT 65413 SIP User s Manual 3 Web Based Management Once you access the Multiple Interface Table page the IP Settings page is no longer available For a detailed description with examples for configuring multiple network interfaces refer to Network Configuration on page 448 You can view all configured IP interfaces that are currently active in the IP Active Interfaces page refer to Viewing Active IP Interfaces on page 174 When adding more than one interface to the table ensure that you enable VLANs using the VLAN Mode VIANMode parameter When booting using BootP DHCP protocols refer to the Product Reference Manual an IP address is obtained from the server This address is used as the OAMP address for this session overriding the IP address you configured in the Multiple Interface Table page The address specified in this table takes effect only after you
217. Index Dns2lp_DomainName Dns2lp_FirstIpAddress Dns2lp_SecondlpAddress Dns2lp_ThirdlipAddress Dns2lp_FourthlpAddress Dns2Ip For example Dns2lp 0 DnsName 1 1 1 1 2 2 2 2 3 3 3 3 4 4 4 4 Notes This parameter can include up to 20 indices If the internal DNS table is used the device first attempts to resolve a domain name using this table If the domain name isn t found the device performs a DNS resolution using an external DNS server To configure the internal DNS table using the Web interface and for a description of the parameters in this ini file table parameter refer to Configuring the Internal DNS Table on page 134 For an explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 This ini file table parameter defines the internal SRV table for resolving host names into DNS A Records Three different A Records can be assigned to a host name Each A Record contains the host name priority weight and port The format of this parameter is as follows SRV2IP FORMAT SRV2IP Index SRV2IP_InternalDomain SRV2IP_TransportType SRV2IP_Dns1 SRV2IP_Priority1 SRV2IP_Weight1 SRV2IP_Port1 SRV2IP_Dns2 SRV2IP_Priority2 218 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Version 6 0 Description SRV2IP_Weight2 SRV2IP_Port2 SRV2IP_Dns3 SRV2IP_Priority3 SRV2IP_Weight3 SRV2IP_Port3 SRV2IP
218. Interface Name Column This column allows the configuration of a short string up to 16 characters to name this interface This name is displayed in management interfaces Web CLI and SNMP and is used in the Media Realm table This column must have a unique value for each interface no two interfaces can have the same name and must not be left blank 10 8 1 3 Other Related Parameters The Multiple Interface table allows you to configure interfaces and their related parameters such as their VLAN ID or the interface name This section lists additional parameters complementing this table functionality 10 8 1 3 1Booting using DHCP The DHCPEnable parameter enables the device to boot while acquiring an IP address from a DHCP server Note that when using this method Multiple Interface table VLANs and other advanced configuration options are disabled 10 8 1 3 2Enabling VLANs The Multiple Interface table s column VLAN ID assigns a VLAN ID to each of the interfaces Incoming traffic tagged with this VLAN ID are channeled to the related interface and outgoing traffic from that interface are tagged with this VLAN ID When VLANs are required the parameter should be set to 1 The default value for this parameter is 0 disabled 10 8 1 3 3 Native VLAN ID A Native VLAN ID is the VLAN ID to which untagged incoming traffic are assigned Outgoing packets sent to this VLAN are sent only with a priority tag VLAN ID 0 When the Native V
219. LAN ID is equal to one of the VLAN IDs configured in the Multiple Interface table and VLANs are enabled untagged incoming traffic are considered as an incoming traffic for that interface Outgoing traffic sent from this interface are sent with the priority tag tagged with VLAN ID 0 When the Native VLAN ID is different from any value in the VLAN ID column in the Multiple Interface table untagged incoming traffic are discarded and all the outgoing traffic are fully tagged SIP User s Manual 452 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities The Native VLAN ID is configurable using the VlanNativeVlanld parameter refer to the Setting up your System sub section below The default value of the Native VLAN ID is 1 If VlanNative Vlanld is not configured i e its default value of 1 occurs but one of the interfaces has a VLAN ID configured to 1 this interface is still related to the Native VLAN If you do not wish to have a Native VLAN ID and want to use VLAN ID 1 ensure that the value of the VlanNative Vlanld parameter is different than any VLAN ID in the table 10 8 1 3 4Quality of Service Parameters The device allows you to specify values for Layer 2 and Layer 3 priorities by assigning values to the following service classes Network Service class network control traffic ICMP ARP Premium Media service class used for RTP Media traffic Premium Control Service class used for Call
220. M 3 5 2 4 Viewing Registration A NAM 3 5 25 Viewing IP COnmectlty soca ccc sccieccsscisiecctec cassacecssecscsasconstans cxoseeisscsstnanen 1 Be 4 INI File Configuration asc sessing kctsaintanntscntncnnccnc NAE Va 4 1 INI File Format er EI A AR ache kiki ben sokn keer TO 4 1 1 Goia Individual ie ini i File Pa a ft 4 1 2 Configuring ini File Table Parame 186 4 1 3 General ini File Formatting Rules o LO 4 2 Modifying an ini Peasia E E E a s bk kajnk 188 SIP User s Manual 4 Document LTRT 65413 SIP User s Manual Contents 4 3 Secured Encoded ini File 189 C 192 5 3 Addin K aaa n ld n u E kien p na ee 5 10 esk the Device 6 Configuration Parameters Reference cs ccccsssceesseeeeeeeeeeeseeeeeeeeeeeeees 207 6 1 6 5 RADIUS Pena PEREM 66 SNMP Paraiba cutl k ako ok k ode la a akt o shaz A Version 6 0 5 March 2010 8 7a c AudioCodes MediaPack Series 67 SIP Configuration Fara saa ca crests erect deco ccetnnotaterasreeeene 6 7 1 General SIP Parame P s EAEE 6 7 2 IP Group Proxy Registration id Authe tio rameters s 203 6 7 3 Voice Mail Parameters APE UP CAE E PAR 274 6 7 4 Fax and Modem Paramete 6 7 5 DTMF and Hook Flash Paramete ae ato A TEE AAE EEA A TANASE ero 6 7 6 Digit Collection and Dial Plan Parameters E PEE PE T EET E 6 7 7 Coders and Profile Parameters s 284 6 8 Supplementary Services ae R ATE ENE 6 8 1
221. March 2010 7a E tall AudioCodes MediaPack Series e DJBufOptFactor e EnableSilenceCompression EnableEchoCanceller Note This mode can be used for fax but is not recommended for modem transmission Instead use the modes Bypass refer to Fax Modem Bypass Mode on page 403 or Transparent with Events refer to Fax Modem Transparent with Events Mode on page 405 for modem 9 6 2 8 RFC 2833 ANS Report upon Fax Modem Detection The device terminator gateway sends RFC 2833 ANS ANSam events upon detection of fax and or modem answer tones i e CED tone This causes the originator to switch to fax modem This parameter is applicable only when the fax or modem transport type is set to bypass Transparent with Events V 152 VBD or G 711 transport When the device is located on the originator side it ignores these RFC 2833 events Relevant parameters m IsFaxUsed 0 or 3 m FaxTransportType 2 E FaxModemNTEMode 1 m VxxModemTransportType 2 9 6 3 V 34 Fax Support V 34 fax machines can transmit data over IP to the remote side using various methods The device supports the following modes for transporting V 34 fax data over IP m Bypass mechanism for V 34 fax transmission refer to Using Bypass Mechanism for V 34 Fax Transmission on page 406 m 138 Version 0 relay mode i e fallback to T 38 refer to Using Relay mode for both T 30 and V 34 faxes on page 407 Using the ini file parameter V34FaxTranspor
222. MediaPack Series 6 46 IPSec Parameters The Internet Protocol security IPSec parameters are described in the table below Table 6 23 IPSec Parameters Parameter Description IPSec Parameters Web Enable IP Security Enables or disables IPSec on the device EMS IPSec Enable f EnablelPSec 0 Disable default 1 Enable Note For this parameter to take effect a device reset is required Web IP Security Associations Table EMS IPSec SA Table IPSecSATable This ini file table parameter configures the IPSec SA table This table allows you to configure the Internet Key Exchange IKE and IP Security IPSec protocols You can define up to 20 IPSec peers The format of this parameter is as follows IPsecSATable FORMAT IPsecSATable Index IPsecSATable RemoteEndpointAddressOrName IPsecSATable AuthenticationMethod IPsecSATable Sharedkey IPsecSATable SourcePort IPsecSATable DestPort IPsecSATable Protocol IPsecSATable Phase1SaLifetimelnSec IPsecSATable Phase2SaLifetimelnSec IPsecSATable Phase2SaLifetimelnKB IPsecSATable DPDmode IPsecSATable IPsecMode IPsecSATable Remote TunnelAddress IPsecSATable RemoteSubnetlPAddress IPsecSATable RemoteSubnetPrefixLength IPsecSATable For example IPsecSATable 1 0 10 3 2 73 0 123456789 0 0 0 0 28800 3600 In the above example a single IPSec IKE peer 10 3 2 73 is configured Pre shared key authentication is selected with the pre shared key set t
223. Navigation bar to display a reduced menu tree select the Full option to display all the menus By default the Basic option is selected Figure 3 5 Navigation Tree in Basic and Full View tus Contiguraton Management Dlognoatics Contiguration Management Scenarios Search Scenarios Search Basic Full O Basic Full V Network Settings H network Settings Full Navigation PMedia Settings PMedia Settings Tree View Protocol Configuration PUB security Setting Option i advance Applications t protocol Configuration s H Advance Applications Only Basic Menus All Menus Note When in Scenario mode refer to Scenarios on page 37 the Navigation tree is displayed in Full view i e all menus are displayed in the Navigation tree SIP User s Manual 28 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 5 2 Showing Hiding the Navigation Pane The Navigation pane can be hidden to provide more space for elements displayed in the Work pane This is especially useful when the Work pane displays a page with a table that s wider than the Work pane and to view the all the columns you need to use scroll bars The arrow button located just below the Navigation bar is used to hide and show the Navigation pane m To hide the Navigation pane click the left pointing arrow S the pane is hidden and the button is replaced by the right pointing arrow button m To show the Navigation pane click t
224. OAMP and Media applications are allowed on the interface 4 OAMP Control Only OAMP and Call Control applications are allowed on the interface 5 Media Control Only Media and Call Control applications are allowed on the interface 6 OAMP Media Control All application types are allowed on the interface Notes A single OAMP interface and only one must be SIP User s Manual 52 Document LTRT 65413 SIP User s Manual Parameter Web EMS IP Address InterfaceTable_IPAddres Web EMS Prefix Length InterfaceTable_PrefixLength Version 6 0 3 Web Based Management Description configured This OAMP interface can be combined with Media and Control Atleast one interface with Media and at least one interface with Control must be configured Multiple interfaces for Media Control and Media and Control can be configured Atleast one IPv4 interface with Control must be configured This can be combined with OAMP and Media Atleast one IPv4 interface with Media must be configured This can be combined with OAMP and Control The IPv4 IP address in dotted decimal notation Notes Each interface must be assigned a unique IP address When booting using BootP DHCP protocols an IP address is obtained from the server This address is used as the OAMP address for the initial session overriding the address configured using the InterfaceTable The address configured for OAMP applicati
225. OR registration record in the device s database The INVITE is then sent to the IP address of the registered contact The default is 1 Note This parameter is only relevant if the parameter Destination Type is set to IP Group However regardless of the settings of the parameter Destination Type the IP Group is still used only for determining the IP Profile 147 March 2010 A c tal AudioCodes MediaPack Series Parameter Description Destination Address The destination IP address or domain name e g IP2IPRouting_DestAddress domain com to where the call is sent Notes This parameter is applicable only if the parameter Destination Type is set to Dest Address 1 When using domain names enter a DNS server IP address or alternatively define these names in the Internal DNS Table refer to Configuring the Internal SRV Table on page 134 Destination Port The destination port to where the call is sent IP2IPRouting_DestPort Destination Transport Type The transport layer type for sending the call IP2IPRouting_DestTransportType 1 Not Configured default 0 UDP 1 TCP 2 TLS Note When this parameter is set to 1 the transport type is determined by the parameter SIPTransportType 3 3 5 Advanced Applications The Advanced Applications menu allows you to configure advanced SIP based applications This menu includes the following page items m Voice Mail Settings re
226. P flow RTP Multiplexing must be enabled on both devices When VLANs are implemented the RTP Multiplexing mechanism is not supported SIP User s Manual 440 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 15 Dynamic Jitter Buffer Operation Voice frames are transmitted at a fixed rate If the frames arrive at the other end at the same rate voice quality is perceived as good In many cases however some frames can arrive slightly faster or slower than the other frames This is called jitter delay variation and degrades the perceived voice quality To minimize this problem the device uses a jitter buffer The jitter buffer collects voice packets stores them and sends them to the voice processor in evenly spaced intervals The device uses a dynamic jitter buffer that can be configured using the following two parameters m Minimum delay DJBufMinDelay 0 msec to 150 msec Defines the starting jitter capacity of the buffer For example at 0 msec there is no buffering at the start At the default level of 10 msec the device always buffers incoming packets by at least 10 msec worth of voice frames m Optimization Factor DJBufOptFactor 0 to 12 13 Defines how the jitter buffer tracks to changing network conditions When set at its maximum value of 12 the dynamic buffer aggressively tracks changes in delay based on packet loss statistics to increase the size of the buffer and doesn t decay back down
227. Phone Context As Prefix AddPhoneContextAsPrefix SIP User s Manual MediaPack Series Description This ini file table parameter manipulates the redirect number for Tel to IP calls The manipulated Redirect Number is sent in the SIP Diversion History Info or Resource Priority headers The format of this parameter is as follows RedirectNumberMapTel2Ip FORMAT RedirectNumberMapTel2Ip Index RedirectNumberMapTel2Ip DestinationPrefix RedirectNumberMapTel2Ip RedirectPrefix RedirectNumberMapTel2Ip NumberType RedirectNumberMapTel2Ip NumberPlan RedirectNumberMapTel2Ip RemoveFromLetft RedirectNumberMapTel2Ip RemoveFromRight RedirectNumberMapTel2Ip LeaveFromRight RedirectNumberMapTel2Ip Prefix2Add RedirectNumberMapTel2Ip Suffix2Add RedirectNumberMapTel2Ip IsPresentationRestricted RedirectNumberMapTel2Ip SrcTrunkGroupID RedirectNumberMapTel2Ip SrclPGroupID RedirectNumberMapTel2Ip For example RedirectNumberMapTel2Ip 1 4 255 255 0 0 255 972 255 1 2 Notes This parameter table can include up to 20 indices 1 20 If the table s matching characteristics rule i e DestinationPrefix RedirectPrefix SrcTrunkGroupID and SrclPGroupID is located for the Tel to IP call then the redirect number manipulation rule defined by the other parameters is applied to the call The manipulation rules are performed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and th
228. Proxies Host names IP addresses and is not marked as critical the Common Name CN of the Subject field is compared with this value If not equal the TLS connection is not established If the CN uses a domain name the certificate can also use wildcards to replace parts of the domain name The valid range is a string of up to 49 characters Note This parameter is applicable only if the parameter PeerHostNameVerificationMode is set to 1 or 2 6 4 5 SSH Parameters The Secure Shell SSH parameters are described in the table below Parameter SSHAdminKey SSHRequirePublicKey Web EMS SSH Server Enable SSHServerEnable Web EMS SSH Server Port SSHServerPort Version 6 0 Table 6 22 SSH Parameters Description Determines the RSA public key for strong authentication to logging in to the SSH interface if enabled The value should be a base64 encoded string The value can be a maximum length of 511 characters For additional information refer to the Product Reference Manual Enables or disables RSA public keys for SSH 0 RSA public keys are optional if a value is configured for the parameter SSHAdminKey default 1 RSA public keys are mandatory Enables or disables the embedded SSH server 0 Disable default 1 Enable Defines the port number for the embedded SSH server Range is any valid port number The default port is 22 237 March 2010 e AudioCodes
229. Record Route SASEnableRecordRoute SIP User s Manual Table 6 46 SAS Parameters Description Enables the Stand Alone Survivability SAS feature 0 Disable Disabled default 1 Enable SAS is enabled When enabled the device receives the registration requests from different SIP entities in the local network and then forwards them to the defined proxy If the connection to the proxy fails Emergency Mode the device serves as a proxy by allowing calls internal to the local network or outgoing to PSTN Note For this parameter to take effect a device reset is required Local UDP port for sending and receiving SIP messages for SAS The SIP entities in the local network need to send the registration requests to this port When forwarding the requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5080 The default gateway used in SAS Emergency Mode When an incoming SIP INVITE is received and the destination Address Of Record is not included in the SAS database the request is immediately sent to this default gateway The address can be configured as an IP address dotted decimal notation or as a domain name up to 49 characters The default is a null string which is interpreted as the local IP address of the gateway Determines the value of the SIP Expires header that is sent ina 200 OK response to an incoming REGISTER message when
230. RedundancyNegotiation Web RFC 2198 Payload Type EMS Redundancy Payload Type RFC2198PayloadType Web Packing Factor EMS Packetization Factor RTPPackingFactor Web EMS Basic RTP Packet Interval BasicRTPPacketinterval Web RTP Directional Control RTPDirectionControl Web EMS RFC 2833 TX Payload Type RFC2833TxPayloadType Web EMS RFC 2833 RX Payload Type RFC2833RxPayloadType EnableDetectRemoteMACChange SIP User s Manual MediaPack Series Description Determines whether the device includes the RTP redundancy dynamic payload type in the SDP according to RFC 2198 0 Disable default 1 Enable When enabled the device includes in the SDP message the RTP payload type RED and the payload type configured by the parameter RFC2198PayloadType a rtpmap lt PT gt RED 8000 Where lt PT gt is the payload type as defined by RFC2198PayloadType The device sends the INVITE message with a rtpmap lt PT gt RED 8000 and responds with a 18x 200 OK and a rtpmap lt PT gt RED 8000 in the SDP Notes For this feature to be functional you must also set the parameter RTPRedundancyDepth to 1 i e enabled Currently the negotiation of RED payload type is not supported and therefore it should be configured to the same PT value for both parties RTP redundancy packet payload type according to RFC 2198 The range is 96 to 127 The default is 104 Note This parameter is a
231. Reference 6 15 Hunt Groups Number Manipulation and Routing Parameters This subsection describes the device s number manipulation and routing parameters 6 15 1 Hunt Groups and Routing Parameters The routing parameters are described in the table below Table 6 54 Routing Parameters Parameter Description Web Endpoint Phone Number Table EMS SIP Endpoints gt Phones TrunkGroup This ini file table parameter is used to define and activate the device s endpoints by defining telephone numbers and assigning them to Hunt Groups The format of this parameter is shown below TrunkGroup FORMAT TrunkGroup Index TrunkGroup TrunkGroupNum TrunkGroup FirstTrunkld TrunkGroup FirstBChannel TrunkGroup LastBChannel TrunkGroup FirstPhoneNumber TrunkGroup Profileld TrunkGroup LastTrunkld TrunkGroup Module TrunkGroup For example the configuration below assigns channels 1 through 4 to Hunt Group 1 and assigns phone numbers 101 to Channel 1 102 to Channel 2 and so on TrunkGroup 0 1 255 1 4 101 0 255 255 Notes The first entry in this table starts at index 0 Each endpoint i e channel must be assigned a unique phone number In other words no two endpoints can have the same phone number The parameters TrunkGroup_FirstTrunkld TrunkGroup_LastTrunkld and TrunkGroup_Module are not applicable For configuring this table in the Web interface refer to Configuring Endpoint Phone Numbers o
232. Routing Index drop down list select the range of entries that you want to edit 3 Configure the Hunt Group according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 The following example shows a REGISTER message for registering endpoint 101 using registration Per Endpoint mode The SipGroupName in the request URI is taken from the IP Group table Version 6 0 85 March 2010 A c tal AudioCodes MediaPack Series REGISTER sip SipGroupName SIP 2 0 Via SIP 2 0 UDP 10 33 37 78 branch z9hG4bKac862428454 From lt sip 101eGatewayName gt tag 1c862422082 To lt sip 101eGatewayName gt CaAll UDe QQHOVO IOS2ASIAOIVO2ZS2SI2 5 10 35 37 78 CSeq 3 REGISTER Contact lt sip 101 10 33 37 78 gt expires 3600 Expires 3600 User Agent Sip Gateway MP 118 FXS FXO v 6 00A 008 002 Content Length 0 Table 3 13 Hunt Group Settings Parameters Parameter Description Hunt Group ID The Hunt Group ID that you want to configure TrunkGroupSettings_TrunkGrou pid Channel Select Mode The method for which IP to Tel calls are assigned to channels TrunkGroupSettings_ChannelSel pertaining to a Hunt Group For a detailed description of this ectMode parameter refer to the global parameter ChannelSelectMode 0 By Dest Phone Number 1 Cyclic Ascending default 2 Ascending 3 Cyclic Descending 4 Descending 5 Dest Nu
233. S Auto Dial Play Busy Tone FXOAutoDialPlayBusyTone Web Hotline Dial Tone Duration EMS Hot Line Tone Duration HotLineToneDuration Web Time Before Reorder Tone sec EMS Time For Reorder Tone TimeBeforeReorderTone Version 6 0 6 Configuration Parameters Reference Description You can configure multiple entries with different source prefixes and tones for the same FXS port Duration in seconds that the dial tone is played FXS interfaces play the dial tone after the phone is picked up off hook FXO interfaces play the dial tone after the port is seized in response to ringing from PBX PSTN The valid range is 0 to 60 The default time is 16 Notes During play of dial tone the device waits for DTMF digits This parameter is not applicable when Automatic Dialing is enabled Duration in msec of the Confirmation tone A Stutter tone is played instead of a regular dial tone when a Message Waiting Indication MWI is received The Stutter tone is composed of a Confirmation tone Tone Type 8 which is played for the defined duration StutterToneDuration followed by a Stutter Dial tone Tone Type 15 Both these tones are defined in the CPT file The range is 1 000 to 60 000 The default is 2 000 i e 2 seconds Notes This parameter is applicable only to FXS interfaces If you want to configure the duration of the Confirmation tone to longer than 16 seconds you must increase the va
234. S interfaces In this configuration the FXO device routes calls received from the PBX to the Remote PBX Extension connected to the FXS device The routing is transparent as if the telephone connected to the FXS device is directly connected to the PBX The following is required m One FXO interfaces with ports connected directly to the PBX lines shown in the figure below One FXS interfaces for the remote PBX extension Analog phones POTS PBX one or more PBX loop start lines LAN network Figure 9 8 FXO FXS Remote PBX Extension Example FXO Device FXS Device 10 1 10 2 y 10 1 10 3 PBX Line PBX Line Phone 100 Phone 101 vd none 8101 rom Phone 100 Phone 201 Remote PBX Extensions SIP User s Manual 394 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 4 3 1 Dialing from Remote Extension Phone at FXS The procedure below describes how to dial from the remote PBX extension i e phone connected to the FXS interface gt To make a call from the FXS interface 1 Off hook the phone and wait for the dial tone from the PBX This is as if the phone is connected directly to the PBX The FXS and FXO interfaces establish a voice path connection from the phone to the PBX immediately after the phone is off hooked 2 Dial the destination number e g phone number 201 The DTMF digits are sent over IP directly to the PBX All the audible tones are generated from the PBX such
235. SASProxySet 1 m P2IPRouting SAS call routing rules 9 2 1 2 Configuring SAS Emergency Calls The device s SAS agent can be configured to detect a user defined emergency number e g 911 in North America which it then redirects the call directly to the PSTN through its FXO interface The emergency number is configured using the ini file parameter SASEmergencyNumbers for a detailed description refer to SIP Configuration Parameters on page 245 Figure 9 3 Device s SAS Agent Redirecting Emergency Calls to PSTN IP Centrex VoIP Device with PSTN Network gt FXO SAS Enabled Interface Emergency Calls e g 911 Routed to PSTN T Phones To configure support for emergency calls configure the parameters below The device and the SAS feature are configured independently If the device and the SAS agent use different proxies then the device s proxy server is defined using the Use Default Proxy parameter while the SAS proxy agent is defined using the Proxy Se table and SASProxySet parameter E EnableSAS 1 m SASLocalSIPUDPPort default 5080 m IsProxyUsed 1 m ProxylP 0 lt external proxy IP address device gt E ProxylP 1 lt external proxy IP address SAS gt m IsRegisterNeeded 1 for the device E IsFallbackUsed 0 m SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 m SASDefaultGatewaylIP lt SAS gateway IP add
236. SATable RemoteSubnetIPAddress SIP User s Manual 192 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS IPsecSATable RemoteSubnetPrefixLength IPsecSATable 1 lt IP address gt 0 lt IKE password gt 0 0 0 28800 ASSI00 O O O O 0 0 0 O 0 0 0 16 IPsecSATable EnableIPSec 1 5 2 2 where e lt IKE password gt is the password for the initial IKE pre shared key e lt IP address gt is the IP address of the EMS server used for connecting to the device for which IPSec connectivity is established 5 To end the PuTTY configuration session type a full stop on an empty line the device responds with the following INI File replaced 6 To save the configuration to the non volatile memory type sar the device reboots with IPSec enabled Note If you have enabled IPSec and you want to change the IP address and or IKE password you need to first disable IPSec Perform the procedure as above but omit the lines IPsecSATable and set EnablelPSec to 0 Once you have done this repeat the exact procedure as described above but with the new IP address and or password Changing SSH Login Password For security it is recommended to change the default SSH Client login password using the SSH client gt To change the SSH login password 1 Open an SSH Client session e g PuTTY and then connect using the default user name and password Admin
237. SSL TLS handshakes always start with SSL 2 0 and switch to TLS 1 0 if both peers support it When set to 1 TLS 1 0 is the only version supported clients attempting to contact the device using SSL 2 0 are rejected Note For this parameter to take effect a device reset is required Defines the time interval in minutes between TLS Re Handshakes initiated by the device The interval range is 0 to 1 500 minutes The default is 0 i e no TLS Re Handshake Determines the device s behavior when acting as a server for TLS connections 0 Disable The device does not request the client certificate default 1 Enable The device requires receipt and verification of the client certificate to establish the TLS connection Notes For this parameter to take effect a device reset is required The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName Determines whether the device verifies the Subject Name of a remote certificate when establishing TLS connections 0 Disable Disable default 1 Server Only Verify Subject Name only when acting as a server for the TLS connection 2 Server 8 Client Verify Subject Name when acting as a server or client for the TLS connection When a remote certificate is received and this parameter is not disabled the value of SubjectAltName is compared with the list of available Proxies If a match is found for any of th
238. TE requests are cached This prevents a mixture of REGISTER and INVITE authorizations 2 Full Caches all challenges from the proxies Note Challenge Caching is used with all proxies and not only with the active one 269 March 2010 ca AudioCodes Parameter Web Proxy IP Table EMS Proxy IP ProxylP Web Proxy Set Table EMS Proxy Set ProxySet SIP User s Manual MediaPack Series Description This ini file table parameter configures the Proxy Set table with up to six Proxy Set IDs each with up to five Proxy server IP addresses or fully qualified domain name FQDN Each Proxy Set can be defined with a transport type UDP TCP or TLS The format of this parameter is as follows ProxylP FORMAT Proxylp Index Proxylp IpAddress Proxylp TransportType Proxylp ProxySetld ProxylP For example Proxylp 0 10 33 37 77 1 0 Proxylp 1 10 8 8 10 0 2 Proxylp 2 10 5 6 7 1 1 Notes This parameter can include up to 30 indices 0 29 The Proxy Set represents the destination of the call For assigning various attributes such as Proxy Load Balancing per Proxy Set ID use the parameter ProxySet For configuring the Proxy Set ID table using the Web interface and for a detailed description of the parameters of this ini file table refer to Configuring the Proxy Sets Table on page 97 Foran explanation on using ini file table parameters refer to Configuring ini File Table Pa
239. TROL MEDIA is missing in the IPv4 interfaces m There are too many interfaces with Application Types of OAMP Only one interface SIP User s Manual 456 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities defined but the Application Types column is not set to O M C numeric value 6 An IPv4 interface was defined with Interface Type different than IPv4 Manual 10 Gateway column is filled in more than one row of the same address family Gateway is defined in an interface not having MEDIA as one of its Application Types Two interfaces have the exact VLAN ID value while VLANs are enabled Two interfaces have the same name Two interfaces share the same address space or subnet Apart from these validation errors connectivity problems may be caused by one of the following Trying to access the device with VLAN tags while booting from BootP DHCP Trying to access the device with untagged traffic when VLANS are on and Native VLAN is not configured properly Routing Table is not configured properly 10 8 2 Routing Table The routing table allows you to configure routing rules You may define up to 25 different routing rules using the ini file Web interface and SNMP 10 8 2 1 Routing Table Overview The Routing Table consists of the following Destination 201 201 0 0 202 202 0 0 203 203 0 0 225 225 0 0 Table 10 8 Routing Table Layout Prefix Length Subnet Mask Gateway Interface Metric
240. The Charge Codes Table page is available only for FXS interfaces You can also configure the Charge Codes table using the ini file table parameter ChargeCode gt To configure the Charge Codes table 1 Access the Charge Codes Table page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Charge Codes page item Alternatively you can also access this page from the Metering Tones page refer to Configuring Metering Tones on page 112 Figure 3 71 Charge Codes Table Page v Table Index 0 4 v Time Period 1 Time Period 2 Time Period 3 Time Period 4 Index End Pulse Pulses Eng pulse Pulses Eng pulse Pulses End Pulse nee Time Interval Time Interval Time Interval Time Interval Answer Answer Answer Answer 0 07 30 1 14 20 2 20 15 1 00 60 1 05 60 1 14 20 1 00 60 1 00 60 1 2 Define up to 25 different charge codes each charge code is defined per row Each charge code can include up to four different time periods in a day 24 hours Each time period is composed of the following e The end of the time period in a 24 rounded hour s format e The time interval between pulses in tenths of a second e The number of pulses sent on answer The first time period always starts at midnight 00 It is mandatory that the last time period of each rule ends at midnight 00 This prevents undefined time frames in a day The devic
241. The default is Default Passw Note Instead of configuring this parameter the Authentication table can be used refer to Authentication on page 136 Cnonce string used by the SIP server and client to provide mutual authentication The value is free format i e Cnonce 0a4f113b The default is Default_Cnonce Determines the device s mode of operation when Authentication and Key Agreement AKA Digest Authentication is used 0 Optional Incoming requests that don t include AKA authentication information are accepted default 1 Mandatory Incoming requests that don t include AKA authentication information are rejected Determines the mode for Challenge Caching which reduces the number of SIP messages transmitted through the network The first request to the Proxy is sent without authorization The Proxy sends a 401 407 response with a challenge This response is saved for further uses A new request is re sent with the appropriate credentials Subsequent requests to the Proxy are automatically sent with credentials calculated from the saved challenge If the Proxy doesn t accept the new request and sends another challenge the old challenge is replaced with the new one 0 None Challenges are not cached Every new request is sent without preliminary authorization If the request is challenged a new request with authorization data is sent default 1 INVITE Only Challenges issued for INVI
242. This method is slightly less reliable than the previous one You can use the CPTWizard described in the Reference Manual to analyze Call Progress Tones generated by any PBX or telephone network Relevant parameters DisconnectOnBusyTone and DisconnectOnDialTone m Detection of silence The call is disconnected after silence is detected on both call directions for a specific configurable amount of time The call isn t disconnected immediately therefore this method should only be used as a backup option Relevant parameters EnableSilenceDisconnect and FarEndDisconnectSilencePeriod m Special DTMF code A digit pattern that when received from the Tel side indicates to the device to disconnect the call Relevant ini file parameter TelDisconnectCode m Interruption of RTP stream Relevant parameters BrokenConnectionEventTimeout and DisconnectOnBrokenConnection Note This method operates correctly only if silence suppression is not used SIP User s Manual 392 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities m Protocol based termination of the call from the IP side Note The implemented disconnect method must be supported by the CO or PBX 9 4 2 3 2 Call Termination before Call Establishment The device supports the following call termination methods before a call is established m Call termination upon receipt of SIP error response in Automatic Dialing mode By default when the FXO device
243. To view the IP to Tel and Tel to IP Call Counters pages m Open the Call Counters page that you want to view Status amp Diagnostics tab gt Gateway Statistics menu gt IP to Tel Calls Count or Tel to IP Calls Count page item the figure below shows the IP to Tel Calls Count page Figure 3 114 Calls Count Page vw Number of Attempted Calls Number of Established Calls Percentage of Successful Calls ASR 73 684211 Number of Calls Terminated due to a Busy Line 2 Number of Calls Terminated due to No Answer Number of Calls Terminated due to Forward Number of Failed Calls due to No Route __ Number of Failed Calls due to No Matched Capabilities Number of Failed Calls due to No Resources Number of Failed Calls due to Other Failures Average Call Duration ACD sec Attempted Fax Calls Counter Successful Fax Calls Counter oOoloMNlIolololol lolo SIP User s Manual 178 Document LTRT 65413 SIP User s Manual Counter Number of Attempted Calls Number of Established Calls Percentage of Successful Calls ASR Number of Calls Terminated due to a Busy Line Number of Calls Terminated due to No Answer Number of Calls Terminated due to Forward Number of Failed Calls due to No Route Number of Failed Calls due to No Matched Capabilities Version 6 0 3 Web Based Management Table 3 33 Call Counters Description Description Indicates the numb
244. Toconfigure the keypad features 1 Open the Keypad Features page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Keypad Features page item Figure 3 72 Keypad Features Page v Forward Unconditional No Answer On Busy On Busy or No Answer Do Not Disturb Deactivate Caller ID Restriction Activate Deactivate wv Hotline Activate Deactivate w Transfer Blind w Call Waiting Activate Deactivate wv Reject Anonymous Call Activate Deactivate SIP User s Manual 114 Document LTRT 65413 SIP User s Manual 3 Web Based Management Configure the keypad features as required For a description of these parameters refer to Configuration Parameters Reference on page 207 Click the Submit button to save your changes To save the changes to the flash memory refer to Saving Configuration on page 161 3 3 4 7 Manipulation Tables The Manipulation Tables submenu allows you to configure number manipulation and mapping of NPI TON to SIP messages This submenu includes the following items General Settings refer to Configuring General Settings on page 115 Manipulation tables refer to Configuring the Number Manipulation Tables on page 115 e Dest Number IP gt Tel e Dest Number Tel gt IP e Source Number IP gt Tel e Source Number Tel gt IP Redirect Number Tel gt IP r
245. Type 0 UDP 1 TCP 2 TLS 1 Undefined Note If no transport type is selected the value of the global parameter SIPTransportType is used refer to Configuring SIP General Parameters on page 88 Version 6 0 99 March 2010 ca AudioCodes Parameter Web Proxy Load Balancing Method EMS Load Balancing Method ProxyLoadBalancingMethod Web EMS Enable Proxy Keep Alive EnableProxyKeepAlive SIP User s Manual MediaPack Series Description Enables the Proxy Load Balancing mechanism per Proxy Set ID 0 Disable Load Balancing is disabled default 1 Round Robin Round Robin 2 Random Weights Random Weights When the Round Robin algorithm is used a list of all possible Proxy IP addresses is compiled This list includes all IP addresses per Proxy Set after necessary DNS resolutions including NAPTR and SRV if configured After this list is compiled the Proxy Keep Alive mechanism according to parameters EnableProxyKeepAlive and ProxyKeepAliveTime tags each entry as offline or online Load balancing is only performed on Proxy servers that are tagged as online All outgoing messages are equally distributed across the list of IP addresses REGISTER messages are also distributed unless a RegistrarlP is configured The IP addresses list is refreshed according to ProxylPListRefresh Time If a change in the order of the entries in the list occurs all load statistics ar
246. VITE and responded to either in the 183 response for early dialogs or in the 200 OK response for confirmed dialogs For outgoing calls Tel to IP the request may be received in the 183 for early dialogs and responded to in the PRACK or received in the 200 OK for confirmed dialogs and responded to in the ACK 2 Once the device receives such a request it sends a SIP response message using the X Detect header to the remote party listing all supported events that can be detected The absence of the X Detect header indicates that no detections are available 3 Each time the device detects a supported event the event is notified to the remote party by sending an INFO message with the following message body e Content Type application X DETECT e Type CPT FAX PTT e Subtype xxx according to the defined subtypes of each type SIP User s Manual 434 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities Below is an example of SIP messages using the X Detect header INVITE sip 101 10 33 2 53 user phone SIP 2 0 Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 Max Forwards 70 From anonymous lt sip anonymouseanonymous invalid gt tag 1c25298 To lt sip 101 10 33 2 53 user phone gt Callil i1pg UL 3342 53 CSeq 1 INVITE Contact lt sip 100 10 33 2 53 gt X Detect Request CPT FAX SIP 2 0 200 OK Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 From anonymous lt sip anony
247. Web interface refer to Configuring the Number Manipulation Tables on page 115 Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Destination Phone Number Manipulation Table for IP to Tel Calls EMS EMS SIP Manipulations gt Destination IP to Telcom NumberMapIP2Tel SIP User s Manual This ini file table parameter manipulates the destination number of IP to Tel calls The format of this parameter is as follows NumberMaplp2Tel FORMAT NumberMaplp2Tel Index NumberMaplp2Tel DestinationPrefix NumberMaplp2Tel SourcePrefix NumberMaplp2Tel_ SourceAddress NumberMaplp2Tel NumberType NumberMaplp2Tel NumberPlan NumberMaplp2Tel_ RemoveFromLeft NumberMaplp2Tel RemoveFromRight NumberMaplp2Tel LeaveFromRight NumberMaplp2Tel Prefix2Add NumberMaplp2Tel Suffix2Add NumberMaplp2Tel IsPresentationRestricted NumberMaplp2Tel For example NumberMaplp2Tel 0 03 22 2 667 Notes This table parameter can include up to 100 indices The parameter IsPresentationRestricted is not applicable The parameters SrclPGroupID NumberType and NumberPlan are not applicable RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add and LeaveFromRight are applied if the called and calling numbers match the DestinationPrefix SourcePrefix and SourceAddress conditions The manipulation rules are executed in the following order RemoveFro
248. When this parameter is set to 2 the XferPrefix parameter can be used to define different routing rules for redirected Calls This parameter is ignored if the parameter AlwaysSendToProxy is set to 1 3 3 4 4 2 Configuring the Account Table The Account Table page allows you to define accounts per Hunt Group Served Hunt Group for registration and or digest authentication user name and password to a destination IP address Serving IP Group The Account table can be used for example to register to an Internet Telephony Service Provider ITSP on behalf of an IP PBX to which the device is connected The registrations are sent to the Proxy Set ID refer to Configuring the Proxy Sets Table on page 97 associated with these Serving IP Groups A Hunt Group can register to more than one Serving IP Group e g ITSP s This can be achieved by configuring multiple entries in the Account table with the same Served Hunt Group but with different Serving IP Groups user name password host name and contact user values Note You can also configure the Account table using the ini file table parameter Account refer to SIP Configuration Parameters on page 245 gt To configure Accounts 1 Open the Account Table page Configuration tab gt Protocol Configuration menu gt Proxies Registration IP Groups submenu gt Account Table page item Figure 3 61 Account Table Page Index Served Trunk Group Serving IP Group Use
249. XS endpoint s Typically the Ringing and or Call Waiting tone played is indicated in the SIP Alert info header field of the received INVITE message If this header is not present in the received INVITE then this feature is used and the tone played is according to the settings in this table For example to configure Distinctive Ringing and Call Waiting tones of Index 9 in the CPT file for FXS endpoints 1 to 4 when a call is received from a source number with prefix 2 configure the following in the ini file ToneIndex FORMAT ToneIndex Index ToneIndex FXSPort First ToneIndex FXSPort Last ToneIndex SourcePrefix ToneIndex_PriorityIndex Meme tiacies Mace O 3 2 dy ToneIndex FirstCallWaitingToneID 8 8 2 Note that the Call Waiting tone index priority index FirstCallWaitingTonelD For example if you want to select the Call Waiting tone defined in the CPT file at Index 9 then you can enter 1 as the priority index and the value 8 for FirstCallWaitingTonelD The summation of these values equals 9 i e index 9 Prerecorded Tones File The CPT file mechanism has several limitations such as a limited number of predefined tones and a limited number of freguency integrations in one tone To overcome these limitations and provide tone generation capability that is more flexible the Prerecorded Tones PRT file can be used If a specific prerecorded tone exists in the PRT file it takes precedence over the
250. abilities The flowchart above describes the following double call hold scenario 1 A calls B and establishes a voice path A places B on hold A hears a Dial tone and B hears a Held tone A calls C and establishes a voice path B places A on hold B hears a Dial tone B calls D and establishes a voice path A ends call with C A hears a Held tone B ends call with D M 4 ee oS ee B retrieves call with A If a party that is placed on hold e g B in the above example is called by another party e g D then the on hold party receives a Call Waiting tone instead of the Held tone While in a Double Hold state placing the phone on hook disconnects both calls i e call transfer is not performed 9 7 2 Call Pickup The device supports the Call Pick Up feature whereby the FXS user can answer someone else s telephone call by pressing a user defined sequence of phone keys When the user dials the user defined digits e g 77 the incoming call from the other phone is forwarded to the FXS user s phone This feature is configured using the parameter KeyCallPickup Note The Call Pick Up feature is supported only for FXS endpoints pertaining to the same Hunt Group ID 9 7 3 Consultation Feature The device s Consultation feature allows you to place one number on hold and consult privately with another party m The Consultation feature is relevant only for the holding party m After holding a call by pressing
251. able Parameters Parameter Description Matching Characteristics Source Username Prefix The prefix of the user part of the incoming INVITE s source IP2IPRouting_SrcUsernamePrefix URI usually the From URI The default is Note The prefix can be a single digit or a range of digits For available notations refer to Dialing Plan Notation for Routing and Manipulation on page 377 Source Host The host part of the incoming SIP INVITE s source URI IP2IPRouting_SrcHost usually the From URI If this rule is not required leave the field empty To denote any host name use the asterisk symbol The default is Destination Username Prefix The prefix of the incoming SIP INVITE s destination URI IP2IPRouting DestUsernamePrefix usually the Request URI user part If this rule is not reguired leave the field empty To denote any prefix use the asterisk symbol The default is SIP User s Manual 146 Document LTRT 65413 SIP User s Manual Parameter Destination Host IP2IPRouting_DestHost 3 Web Based Management Description The host part of the incoming SIP INVITE s destination URI usually the Request URI If this rule is not required leave the field empty The asterisk symbol can be used to depict any destination host The default is Operation Routing Rule performed when match occurs in above characteristics Destination Type IP2IPRouting_DestType Destination IP Group
252. able only to FXS interfaces Web Emergency Calls Determines the time in minutes that the device waits before tearing Regret Timeout down an emergency call defined by the parameter EMS Emergency Regret EmergencyNumbers Until this time expires an emergency call can Timeout only be disconnected by the remote party typically by a Public EmergencyRegretTimeout Safety Answering Point PSAP The valid range is 1 to 30 The default value is 10 Note This parameter is applicable only to FXS interfaces 6 8 9 FXS Call Cut Through Parameter The FXS off hook call cut through parameter is described in the table below Table 6 42 Call Cut Through Parameter Parameter Description Web Enable Calls Cut Enables FXS endpoints to receive incoming IP calls while the port is in an Through off hook state EMS Cut Through fault CutThrough 0 Disable default 1 Enable If enabled the FXS interface answers the call and cuts through the voice channel if there is no other active call on the port even if the port is in off hook state When the call is terminated by the remote party the device plays a reorder tone for a user defined time configured by the parameter TimeForReorderTone and is then ready to answer the next incoming call without on hooking the phone The waiting call is automatically answered by the device when the current call is terminated configured by setting the parameter EnableCallWaiting to 1
253. ach endpoint registers and authenticates separately with its own user name and password Single registration and authentication Authentication Mode 1 is usually defined for FXO ports Enables setting an endpoint or the entire device i e all endpoints to out of service if registration fails 0 Disable default 1 Enable If the registration is per endpoint i e AuthenticationMode is set to 0 or per Account refer to Configuring Hunt Group Settings on page 85 and a specific endpoint Account registration fails SIP 4xx or no response then that endpoint is set to out of service until a success response is received in a subsequent registration request When the registration is per the entire device i e AuthenticationMode is set to 1 and registration fails all endpoints are set to out of service Note Te out of service method is configured using the parameter FXSOOSBehavior Determines whether the device performs an explicit unregister 0 Disable default 1 Enable The device sends an asterisk value in the SIP Contact header instructing the Registrar server to remove all previous registration bindings When enabled the device removes SIP User Agent UA registration bindings in a Registrar according to RFC 3261 Registrations are soft state and expire unless refreshed but they can also be explicitly removed A client can attempt to influence the expiration interval selected by the Registrar
254. affic to send it never sends a DPD message Note For detailed information on DPD refer to the Product Reference Manual Defines the IP address of the peer router Note This parameter is applicable only if the Operational Mode is set to Tunnel Defines the IP address of the remote subnet Together with the Prefix Length parameter below this parameter defines the network with which the IPSec tunnel allows communication Note This parameter is applicable only if the Operational Mode is set to Tunnel 82 Document LTRT 65413 SIP User s Manual Parameter Name Remote Prefix Length IPsecSATable_RemoteSubnetPrefix Length Version 6 0 3 Web Based Management Description Defines the prefix length of the Remote Subnet IP Address parameter in bits The prefix length defines the subnet class of the remote network A prefix length of 16 corresponds to a Class B subnet 255 255 0 0 a prefix length of 24 corresponds to a Class C subnet 255 255 255 0 Note This parameter is applicable only if the Operational Mode is set to Tunnel 83 March 2010 7a c tal AudioCodes MediaPack Series 3 3 4 Protocol Configuration The Protocol Configuration menu allows you to configure the device s SIP parameters and contains the following submenus m Applications Enabling refer to Enabling Applications on page 84 Hunt Group refer to Hunt Group on page 85 Protocol Definition refer to Protocol Defin
255. ain IP route above any alternative route in the table When an appropriate entry destination number matches one of the prefixes is found the prefix s corresponding destination IP address is verified If the destination IP address is disallowed or if the original call fails and the device has made two additional attempts to establish the call without success an alternative route is searched in the table and used for routing the call Destination IP address is disallowed if no ping to the destination is available ping is continuously initiated every seven seconds when an inappropriate level of QoS was detected or when a DNS host name is not resolved The QoS level is calculated according to delay or packet loss of previously ended calls If no call statistics are received for two minutes the QoS information is reset 9 5 2 Determining the Availability of Destination IP Addresses To determine the availability of each destination IP address or host name in the routing table one or all of the following user defined methods are applied m Connectivity The destination IP address is queried periodically currently only by ping m QoS The QoS of an IP connection is determined according to RTCP statistics of previous calls Network delay in msec and network packet loss in percentage are separately quantified and compared to a certain configurable threshold If the calculated amounts of delay or packet loss exceed these thresholds
256. al INVITE used to establish the voice call not fax was already sent a CANCEL if not connected yet or a BYE if already connected is sent to tear down the voice call Notes To enable this feature set the parameter CNGDetectorMode to 2 and the parameter IsFaxUsed to 1 2 or 3 The FAX prefix in routing and manipulation tables is case sensitive Determines the device s behavior upon detection of a CNG tone 0 Does not send a SIP Re INVITE upon detection of a fax CNG tone when the parameter CNGDetectorMode is set to 1 default 1 Sends a SIP Re INVITE upon detection of a fax CNG tone when the parameter CNGDetectorMode is set to 1 Determines when the device initiates a T 38 session for fax transmission 0 Initiate T 38 on Preamble The device to which the called fax is connected initiates a T 38 session on receiving HDLC Preamble signal from the fax default 1 Initiate T 38 on CED The device to which the called fax is connected initiates a T 38 session on receiving a CED answer tone from the fax This option can only be used to relay fax signals as the device sends T 38 Re INVITE on detection of any fax modem Answer tone 2100 Hz amplitude modulated 2100 Hz or 2100 Hz with phase reversals The modem signal fails when using T 38 for fax relay Notes For this parameter to take effect a device reset is required This parameters is applicable only if the parameter IsFaxUsed is set to 1 T
257. aled numbers that begin with 00 and then any digit from 1 through 7 followed by three digits of any number Version 6 0 379 March 2010 e AudioCodes MediaPack Series If you want the device to accept dial any number ensure that the digit map contains the rule x T otherwise dialed numbers not represented in the digit map are rejected If an external Dial Plan is implemented for dialing plans refer to External Dial Plan File on page 380 then digit mapping configured by the parameter DigitMapping is ignored 9 13 External Dial Plan File The device allows you to select a specific Dial Plan index defined in an external Dial Plan file This file is loaded to the device as a dat file binary file converted from an ini file using the DConvert utility This file can include up to eight Dial Plans Dial Plan indices The required Dial Plan can be selected using the Dial Plan index using the parameter DialPlanlndex This parameter can use values 0 through 7 where 0 denotes PLANT 1 denotes PLAN2 and so on The Dial Plan index can be configured globally or per Tel Profile The Dial Plan file can include up to 8 000 dialing rules lines The format of the Dial Plan index file is as follows m A name in square brackets on a separate line indicates the beginning of a new Dial Plan index m Every line under the Dial Plan index defines a dialing prefix and the number of digits expected to follow that pre
258. all FXS or FXO device Diverted party new destination of the forwarded call FXS or FXO device The served party FXS interface can be configured through the Web interface refer to Configuring Call Forward on page 140 or ini file to activate one of the call forward modes These modes are configurable per endpoint When call forward is initiated the device sends a SIP 302 response with a contact that contains the phone number from the forward table and its corresponding IP address from the routing table or when a proxy is used the proxy s IP address For receiving call forward the device handles SIP 3xx responses for Version 6 0 redirecting calls with a new contact 413 March 2010 7a K tall AudioCodes MediaPack Series 9 7 5 1 9 7 5 2 Call Forward Reminder Ring The device supports the Call Forward Reminder Ring feature for FXS interfaces whereby the device s FXS endpoint emits a short ring burst only if in onhook state when a third party Application Server e g softswitch forwards an incoming call to another destination This is important in that it notifies audibly the FXS endpoint user that a call forwarding service is currently being performed Figure 9 18 Call Forward Reminder with Application Server FXS Gateway FXS Line POTS Phone Application Server The device generates a Call Forward Reminder ring burst to the FXS endpoint each time it receives a SIP NOTIFY message with a
259. ameters which are typically configured separately using their individual global parameters You can later assign these Tel Profile IDs to other elements such as in the Endpoint Phone Number table TrunkGroup parameter Therefore Tel Profiles allow you to apply the same settings of a group of parameters to multiple channels or apply specific settings to different channels The format of this parameter is as follows TelProfile FORMAT TelProfile Index TelProfile ProfileName TelProfile TelPreference TelProfile CodersGroupID TelProfile IsFaxUsed TelProfile JitterBufMinDelay TelProfile JitterBufOptFactor TelProfile IPDiffServ TelProfile SiglPDiffServ TelProfile DtmfVolume TelProfile InputGain TelProfile VoiceVolume TelProfile EnableReversePolarity TelProfile EnableCurrentDisconnect TelProfile EnableDigitDelivery TelProfile EnableEC TelProfile MWlAnalog TelProfile MWIDisplay TelProfile FlashHookPeriod TelProfile EnableEarlyMedia TelProfile ProgressIndicator2IP TelProfile TimeForReorderTone TelProfile EnableDIDWink TelProfile IsTwoStageDial TelProfile DisconnectOnBusyTone TelProfile EnableVoiceMailDelay TelProfile DialPlanlndex TelProfile Enable911PSAP TelProfile SwapTelTolpPhoneNumbers TelProfile EnableAGC TelProfile ECNIpMode TelProfile For example TelProfile 1 ITSP_audio 1 0 0 10 10 46 40 11 0 0 0 0 0 1 0 0 700 0 1 255 0 1 1 1 1 1 0 0 0 Notes You can con
260. an external line not an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward with no reason when the original call is received from an external line not an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate an internal call The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate an external call The valid range is a 120 character string Determines a digit pattern that when received from the Tel side indicates the device to disconnect the call The valid range is a 25 character string A digit pattern that if received as Src S or Redirect R numbers is ignored and not added to that number The valid range is a 25 character string Fax and Modem Parameters The fax and modem parameters parameters are described in the table below Table 6 30 Fax and Modem Parameters Parameter EMS T38 Use RTP Port T38UseRTPPort Description Defines the port with relation to RTP port for sending and receiving T 38 packets 0 Use the RTP port 2 to send receive T 38 packets default 1 Use the same port as the RTP port to send receive T 38 packets Notes For this parameter to take effect you must reset the device When the device is configured to use V 152 to negotiate audio and T 38 coders the UDP port
261. anagers Disable SNMP Trap Manager Host Name w Activity Types to Report via Activity Log Messages Parameters Value Change 4uxiliary Files Loading Device Reset Flash Memory Burning Device Software Update Access to Restricted Domains Non Authorized Access Sensitive Parameters Value Change 2 Configure the management parameters 3 Configure the following SNMP tables e SNMP Trap Destinations Click the arrow p button to configure the SNMP trap destinations refer to Configuring the SNMP Trap Destinations Table on page 154 e SNMP Community String Click the arrow p button to configure the SNMP community strings refer to Configuring the SNMP Community Strings on page 155 e SNMP V3 Table Click the arrow ue button to configure the SNMP V3 users refer to Configuring SNMP V3 Table on page 156 e SNMP Trusted Managers Click the arrow L button to configure the SNMP Trusted Managers refer to Configuring SNMP Trusted Managers on page 157 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 153 March 2010 7a c tal AudioCodes MediaPack Series 3 4 1 1 1 Configuring the SNMP Trap Destinations Table The SNMP Trap Destinations page allows you to configure up to five SNMP trap managers gt To configure the SNMP Trap Destinations table 1 Access the Management Settings
262. and Root Path of audio For an explanation on configuring Web interface tables refer to Working with Tables on page 34 You can also configure the NFS table using the ini file table parameter NFSServers refer to NFS Parameters on page 216 SIP User s Manual 56 Document LTRT 65413 SIP User s Manual Parameter Index Host Or IP Root Path NFS Version Authentication Type User ID Group ID VLAN Type Version 6 0 3 Web Based Management Table 3 5 NFS Settings Parameters Description The row index of the remote file system The valid range is 1 to 16 The domain name or IP address of the NFS server If a domain name is provided a DNS server must be configured Path to the root of the remote file system in the format path For example audio NFS version used to access the remote file system 2 NFS Version 2 3 NFS Version 3 default Authentication method used for accessing the remote file system 0 Null 1 Unix default User ID used in authentication when using Unix The valid range is 0 to 65537 The default is 0 Group ID used in authentication when using Unix The valid range is 0 to 65537 The default is 1 The VLAN type for accessing the remote file system 0 OAM 1 MEDIA default Note This parameter applies only if VLANs are enabled or if Multiple IPs is configured refer to Network Configuration on page 448 57 March 2010 7a tal
263. ansport Modes The device supports the following transport modes for fax per modem type V 22 V 23 Bell V 32 V 34 m T 38 fax relay refer to T 38 Fax Relay Mode on page 401 m G 711 Transport switching to G 711 when fax modem is detected refer to G 711 Fax Modem Transport Mode on page 402 m Fax fallback to G 711 if T 38 is not supported refer to Fax Fallback on page 402 m Fax and modem bypass a proprietary method that uses a high bit rate coder refer to Fax Modem Bypass Mode on page 403 m NSE Cisco s Pass through bypass mode for fax and modem refer to Fax Modem NSE Mode on page 404 m Transparent with events passing the fax modem signal in the current voice coder with adaptations refer to Fax Modem Transparent with Events Mode on page 405 E Transparent passing the fax modem signal in the current voice coder refer to Fax Modem Transparent Mode on page 405 m RFC 2833 ANS Report upon Fax Modem Detection refer to RFC 2833 ANS Report upon Fax Modem Detection on page 406 Adaptations refer to automatic reconfiguration of certain DSP features for handling fax modem streams differently than voice SIP User s Manual 400 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 6 2 1 1 38 Fax Relay Mode In Fax Relay mode fax signals are transferred using the T 38 protocol T 38 is an ITU standard for sending fax across IP networks in real time mode The device currentl
264. applicable to FXO and FXS interfaces but for FXO the Web interface does not display this parameter It is possible to configure whether the KeyBlindTransfer code is added as a prefix to the dialed destination number by using the parameter KeyBlindTransferAddPrefix 326 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Keypad Feature Call Waiting Parameters Web Activate Keypad sequence that activates the Call Waiting option After EMS Keypad Features CW the sequence is pressed a confirmation tone is heard KeyCallWaiting Web Deactivate Keypad sequence that deactivates the Call Waiting option EMS Keypad Features CW Deact After the sequence is pressed a confirmation tone is heard KeyCallWaitingDeact Keypad Feature Reject Anonymous Call Parameters Web Activate Keypad sequence that activates the reject anonymous call EMS Reject Anonymous Call option whereby the device rejects incoming anonymous calls KeyRejectAnonymousCall After the sequence is pressed a confirmation tone is heard Web Deactivate Keypad sequence that de activates the reject anonymous call EMS Reject Anonymous Call option After the sequence is pressed a confirmation tone is Deact heard KeyRejectAnonymousCallDeact RejectAnonymousCallPerPort This ini file table parameter determines whether the device rejects incoming anonymous calls on FXS interfaces The format of this parameter is
265. ar doesn t respond to the REGISTER message the same REGISTER message is sent immediately to the next Proxy To allow this mechanism the parameter EnableProxyKeepAlive must be set to 0 When a specific transport type is defined using the parameter RegistrarTransportType a DNS NAPTR query is not performed even if the parameter DNSQueryType is set to 2 271 March 2010 ca AudioCodes Parameter Web EMS Registrar Transport Type RegistrarTransportType Web EMS Registration Time RegistrationTime Web Re registration Timing EMS Time Divider RegistrationTimeDivider Web EMS Registration Retry Time RegistrationRetryTime Web Registration Time Threshold EMS Time Threshold RegistrationTimeThreshold Web Re register On INVITE Failure EMS Register On Invite Failure RegisterOnlnviteFailure Web ReRegister On Connection Failure EMS Re Register On Connection Failure ReRegisterOnConnectionFailure SIP User s Manual MediaPack Series Description Determines the transport layer used for outgoing SIP dialogs initiated by the device to the Registrar 1 Not Configured default 0 UDP 1 TCP 2 TLS Note When set to Not Configured the value of the parameter SIPTransportType is used Defines the time interval in seconds for registering to a Proxy server The value is used in the SIP Expires header In addition this parameter defines the time interval between Keep Alive m
266. ard by the called party 15 Stutter Dial Tone 16 Off Hook Warning Tone 17 Call Waiting Ringback Tone heard by the calling party 18 Comfort Tone 23 Hold Tone 46 Beep Tone Tone Modulation Type Amplitude Modulated 1 or regular 0 Tone Form The tone s format can be one of the following Continuous 1 Cadence 2 Burst 3 Low Freq Hz Frequency in Hz of the lower tone component in case of dual frequency tone or the frequency of the tone in case of single tone This is not relevant to AM tones High Freq Hz Frequency in Hz of the higher tone component in case of dual frequency tone or zero 0 in case of single tone not relevant to AM tones Low Freq Level dBm Generation level 0 dBm to 31 dBm in dBm not relevant to AM tones High Freq Level Generation level of 0 to 31 dBm The value should be set to 32 in the case of a single tone not relevant to AM tones 368 Document LTRT 65413 SIP User s Manual 8 Auxiliary Configuration Files e First Signal On Time 10 msec Signal On period in 10 msec units for the first cadence on off cycle For continuous tones this parameter defines the detection period For burst tones it defines the tone s duration e First Signal Off Time 10 msec Signal Off period in 10 msec units for the first cadence on off cycle for cadence tones For burst tones this parameter defines the off time required after the burst tone
267. as follows RejectAnonymousCallPerPort FORMAT RejectAnonymousCallPerPort_Index RejectAnonymousCallPerPort_Enable RejectAnonymousCallPerPort Where Enable accept 0 default or reject 1 incoming anonymous calls For example RejectAnonymousCallPerPort 0 0 RejectAnonymousCallPerPort 1 1 If enabled when a device s FXS interface receives an anonymous call it responds with a 433 Anonymity Disallowed SIP response Notes This parameter is applicable only to FXS interfaces This parameter is per device This parameter can appear up to 8 times for 8 port MP 11x devices and up to 24 times for MP 124 devices Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Version 6 0 327 March 2010 Aa e AudioCodes MediaPack Series 6 13 General FXO Parameters The general FXO parameters are described in the table below Parameter Web FXO Coefficient Type EMS Country Coefficients CountryCoefficients FXONumberOfRings Web EMS Dialing Mode IsTwoStageDial Web EMS Waiting For Dial Tone IsWaitForDialTone SIP User s Manual Table 6 52 General FXO Parameters Description Determines the FXO line characteristics AC and DC according to USA or TBR21 standard 66 Europe TBR21 70 USA United States default Note For this parameter to take effect a device reset is required Defines the number of rings
268. ased Management on page 23 m A configuration ini file loaded to the device refer to ini File Configuration on page 185 m AudioCodes Element Management System refer to Element Management System EMS on page 191 m Simple Network Management Protocol SNMP browser software refer to the Product Reference Manual To initialize the device by assigning it an IP address a firmware file cmp and a configuration file ini file you can use AudioCodes BootP TFTP utility which accesses the device using its MAC address refer to the Product Reference Manual Version 6 0 21 March 2010 A c tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 22 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 Web Based Management The device s Embedded Web Server Web interface provides FCAPS fault management configuration accounting performance and security functionality The Web interface allows you to remotely configure your device for quick and easy deployment including uploading of software cmp configuration ini and auxiliary files and resetting the device The Web interface provides real time online monitoring of the device including display of alarms and their severity In addition it displays performance statistics of voice calls and various traffic parameters The Web interface provides a user friendly graphical user interface GUI which can be accessed using any
269. atchdog Enable Calls Cut Through Enable User Information Usage Out Of Service Behavior Delay After Reset sec T38 Fax Max Buffer Enable Microsoft Extension Reliable Connection Persistent Mode First Call Ringback Tone ID Call Pickup Key Enable Delayed Offer Replace Number Sign With Escape Char IP2IP Registration Time Not Configured v Disable v 0 3 8 0 Disable Disable Disable Reorder Tone 7 1024 Disable Disable 1 Disable Disable 20 wv Emergency Calls Emergency Numbers min Emergency Calls Regret Timeout Configure the parameters as required Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual Document LTRT 65413 110 SIP User s Manual 3 3 4 6 2 Configuring Supplementary Services 3 Web Based Management The Supplementary Services page is used to configure parameters that are associated with supplementary services For a description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 For an overview on supplementary services refer to Working with Supplementary Services on page 409 gt Version 6 0 To configure the supplementary services pa
270. ate 3 In the Lock Timeout field relevant only if the parameter Graceful Option in the previous step is set to Yes enter the time in seconds after which the device locks Note that if no traffic exists and the time has not yet expired the device locks 4 Click the LOCK button a confirmation message box appears requesting you to confirm device Lock Figure 3 103 Device Lock Confirmation Message Box Microsoft Internet Explorer 2 Are you sure you want to Lock the Gateway so incoming calls wil be rejected and active calls will be closed when timeout expires 5 Click OK to confirm device Lock if Graceful Option is set to Yes the lock is delayed and a screen displaying the number of remaining calls and time is displayed Otherwise the lock process begins immediately The Current Admin State field displays the current state LOCKED or UNLOCKED gt To unlock the device 1 Open the Maintenance Actions page refer to Maintenance Actions on page 159 2 Under the LOCK UNLOCK group click the UNLOCK button Unlock starts immediately and the device accepts new incoming calls Version 6 0 161 March 2010 7a L l AudioCodes MediaPack Series 3 4 1 3 3 Saving Configuration The Maintenance Actions page allows you to save burn the current parameter configuration including loaded auxiliary files to the device s non volatile memory i e flash The parameter modifications that y
271. ated For example http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters TLSRootFileUrl Specifies the name of the TLS trusted root certificate file and the URL from where it s downloaded Note For this parameter to take effect a device reset is required Version 6 0 363 March 2010 A K tal AudioCodes MediaPack Series Parameter Description TLSCertFileUrl Specifies the name of the TLS certificate file and the URL from where it s downloaded Note For this parameter to take effect a device reset is required UserInfoFileURL Specifies the name of the User Information file and the path to the server IP address or FQDN on which it is located For example http server_name file https server_name file Note The maximum length of the URL address is 99 characters SIP User s Manual 364 Document LTRT 65413 SIP User s Manual 7 Restoring Factory Default Settings 7 Restoring Factory Default Settings The device provides you with the following methods for restoring the device s configuration to factory default settings m Using the CLI refer to Restoring Defaults using CLI on page 365 m Loading an empty ini file refer to Restoring Defaults using an ini File on page 365 m Using the hardware Reset button refer to Restoring Defaults using Hardware Reset Button on page 366 zl Restoring Defaults using CLI The device can be restored to factory def
272. ates requests and reacts to the resulting responses using the NTP version 3 protocol definitions according to RFC 1305 Through these requests and responses the NTP client synchronizes the system time to a time source within the network thereby eliminating any potential issues should the local system clock drift during operation By synchronizing time to a network time source traffic handling maintenance and debugging become simplified for the network administrator The NTP client follows a simple process in managing system time the NTP client requests an NTP update receives an NTP response and then updates the local system clock based on a configured NTP server within the network The client requests a time update from a specified NTP server at a specified update interval In most situations this update interval is every 24 hours based on when the system was restarted The NTP server identity as an IP address and the update interval are user defined using the ini file parameters NTPServerlP and NTPUpdatelnterval respectively or an SNMP MIB object refer to the Product Reference Manual When the client receives a response to its request from the identified NTP server it must be interpreted based on time zone or location offset that the system is to a standard point of reference called the Universal Time Coordinate UTC The time offset that the NTP client uses is configurable using the ini file parameter NTPServerUTCOffset or v
273. ation sec Hotline Dial Tone Duration sec Enable Special Digits Default Destination Number Special Digit Representation O Basic Full 2 3 Edt Scenario M no TTT ek ome ie il ini i When you select a Scenario Step the corresponding page is displayed in the Work pane In each page the available parameters are indicated by a dark blue background the unavailable parameters are indicated by a gray or light blue background To navigate between Scenario Steps you can perform one of the following E Inthe Navigation tree click the required Scenario Step Version 6 0 39 March 2010 7a K tal AudioCodes MediaPack Series m n an opened Scenario Step i e page appears in the Work pane use the following navigation buttons gt e Next opens the next Step listed in the Scenario 4 Previous opens the previous Step listed in the Scenario Note If you reset the device while in Scenario mode after the device resets you are returned once again to the Scenario mode 3 1 8 3 Editing a Scenario You can modify a Scenario anytime by adding or removing Steps i e pages or parameters and changing the Scenario name and the Steps names Note Only users with access level of Security Administrator can edit a Scenario gt To edit a Scenario 1 On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm Scenario loading 2 Click OK the Scena
274. ations in the current Web session You can also access certain pages from the Device Actions button located on the toolbar refer to Toolbar on page 26 To view all the menus in the Navigation tree ensure that the Navigation tree is in Full view refer to Displaying Navigation Tree in Basic and Full View on page 28 To get Online Help for the currently opened page refer to Getting Help on page 45 Certain pages may not be accessible if your Web user account s access 3 1 6 2 level is low refer to Configuring the Web User Accounts on page 66 Viewing Parameters For convenience some pages allow you to view a reduced or expanded display of parameters A reduced display allows you to easily identify required parameters enabling you to quickly configure your device The Web interface provides you with two methods for handling the display of page parameters m Display of basic and advanced parameters refer to Displaying Basic and Advanced Parameters on page 31 m Display of parameter groups refer to Showing Hiding Parameter Groups on page 32 Note Certain pages may only be read only if your Web user account s access level is low refer to Configuring the Web User Accounts on page 66 If a page is read only Read Only Mode is displayed at the bottom of the page SIP User s Manual 30 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 6 2 1 Displaying Basic and A
275. ative IP address refer to Configuring the Tel to IP Routing on page 126 This call release reason type can be configured for example when there is no response to an INVITE message after INVITE re transmissions the device issues an internal 408 No Response implicit release reason The device also plays a tone to the endpoint whenever an alternative route is used This tone is played for a user defined time configured by the ini file parameter AltRoutingToneDuration SIP User s Manual 124 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To configure the reasons for alternative routing 1 Open the Reasons for Alternative Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Alternative Routing page item Figure 3 77 Reasons for Alternative Routing Page IP to Tel Reasons Reason 1 Reason 2 Reason 3 Reason 4 Tel to IP Reasons Reason 1 Reason 2 Reason 3 Reason 4 2 In the IP to Tel Reasons group select up to four different call failure reasons that invoke an alternative IP to Tel routing 3 In the Tel to IP Reasons group select up to four different call failure reasons that invoke an alternative Tel to IP routing 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 To enable alternative routing using the IP to T
276. atterns in the digit map the device stops collecting digits and establishes a call with the collected number The digit map pattern can contain up to 52 options rules each separated by a vertical bar The maximum length of the entire digit pattern is 152 characters The available notations include the following n m Range of numbers not letters single dot Repeat digits until next notation e g T x Any single digit T Dial timeout configured by the parameter TimeBetweenDigits S Immediately applies a specific rule that is part of a general rule For example if your digit map includes a general rule x T and a specific rule 11x for the specific rule to take precedence 283 March 2010 A c tal AudioCodes MediaPack Series Parameter Web Max Digits in Phone Num EMS Max Digits in Phone Number MaxDigits Web Inter Digit Timeout for Overlap Dialing sec EMS Interdigit Timeout Sec TimeBetweenDigits Description over the general rule append S to the specific rule i e 11xS An example of a digit map is shown below 11xS OOT 1 7 Xxx 8xXxXXXXXx XXXXXXX XX 91XXXXXXXXXX 901 1x T In the example above the last rule can apply to International numbers 9 for dialing tone 011 Country Code and then any number of digits for the local number x Notes Ifthe parameter DialPlanIndex is configured to select a Dial Plan index then the parameter DigitMapping is ig
277. aults using the CLI command RestoreFactorySettings rfs as described in the procedure below gt To restore factory default settings using CLI 1 Access the device s CLI a Connect the device s RS 232 port refer to the Installation Manual to COM1 or COM2 communication port on your PC b Establish serial communication with the device using a serial communication program such as HyperTerminal with the following communication port settings Baud Rate 9 600 bps Data Bits 8 Parity None Stop Bits 1 Flow Control None 2 Atthe CLI prompt enter the following command RestoreFactorySettings 7 2 Restoring Defaults using an ini File You can restore the device s parameters to default settings while retaining its IP address and the Web interface s login user name and password This is achieved by loading an empty ini file to the device The loaded ini file must be empty i e no parameters or have only semicolons preceding all lines When a parameter is absent from a loaded ini file the default value is assigned to that parameter according to the cmp file loaded to the device and saved to the non volatile memory thereby overriding the value previously defined for that parameter Version 6 0 365 March 2010 A e AudioCodes MediaPack Series 7 3 Restoring Defaults using Hardware Reset Button The device s hardware Reset button can be used to reset the device to default settings For a detailed
278. ave the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 150 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 5 3 Configuring FXO Parameters The FXO Settings page allows you to configure the device s specific FXO parameters For a description of these parameters refer to Configuration Parameters Reference on page 207 Note The FXO Settings page is available only for FXO interfaces To configure the FXO parameters 1 Open the FXO Settings page Configuration tab gt Advanced Applications menu gt FXO Settings page item Figure 3 94 FXO Settings Page Dialing Mode Two Stages Waiting for Dial Tone No Time to Wait before Dialing msec 1000 Ring Detection Timeout sec 8 Reorder Tone Duration sec 255 Answer Supervision No Rings before Detecting Caller ID 1 Send Metering Message to IP No Disconnect Call on Busy Tone Detection CAS Enable Disconnect On Dial Tone Disable Guard Time Between Calls n FXO AutoDial Play BusyTone Disable 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 151 March 2010 7a c tal AudioCodes MediaPack Series 3 4 3 4 1 3 4 1 1 Management Tab The Mana
279. ay client certificates Set the parameter Secured Web Connection HTTPS to HTTPS Only 0 in Configuring the General Security Settings on page 78 to ensure you have a method of accessing the device in case the client certificate doesn t work Restore the previous setting after testing the configuration Open the Certificates Signing Reguest page refer to Server Certificate Replacement on page 73 T9 March 2010 A c tal AudioCodes MediaPack Series 3 In the Certificates Files group click the Browse button corresponding to Send Trusted Root Certificate Store file navigate to the file and then click Send File When the operation is complete set the ini file parameter HTTPSRedquireClientCertificates to 1 Save the configuration refer to Saving Configuration on page 161 and then restart the device When a user connects to the secured Web server If the user has a client certificate from a CA that is listed in the Trusted Root Certificate file the connection is accepted and the user is prompted for the system password If both the CA certificate and the client certificate appear in the Trusted Root Certificate file the user is not prompted for a password thus providing a single sign on experience the authentication is performed using the X 509 digital signature If the user doesn t have a client certificate from a listed CA or doesn t have a client certificate at all the connection is re
280. b interface is accessed displaying the Home page for a detailed description of the Home page refer to Using the Home Page on page 47 Note If access to the device s Web interface is denied Unauthorized due to Microsoft Internet Explorer security settings perform the following 1 Delete all cookies in the Temporary Internet Files folder If this does not resolve the problem the security settings may need to be altered continue with Step 2 In Internet Explorer navigate to Tools menu gt Internet Options gt Security tab gt Custom Level and then scroll down to the Logon options and select Prompt for username and password Select the Advanced tab and then scroll down until the HTTP 1 1 Settings are displayed and verify that Use HTTP 1 1 is selected 3 Quit and start the Web browser again SIP User s Manual 24 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 3 Areas of the GUI The figure below displays the general layout of the Graphical User Interface GUI of the Web interface Figure 3 2 Main Areas of the Web Interface GUI K F AudioCodes Microsoft Internet Explorer Ox Fie tat View Favortes Took heb W ck A k ae Boe Toolbar hd Bo 8 hep 20 13 4 13 A Dovce Actions v t Home 6 Hetp S Log on tr be m p ry Status Conthgur ation Managemera Disgnostcs Soananes Seach Bask PoraneterUct a v Syslog Sethngs Basic v Full Syslog Se
281. be set to Using OPTIONS when Proxy redundancy is used When this parameter is set to Using REGISTER the homing 100 Document LTRT 65413 SIP User s Manual Parameter Web Proxy Keep Alive Time EMS Keep Alive Time ProxyKeepAliveTime Web EMS Is Proxy Hot Swap IsProxyHotSwap 3 Web Based Management Description redundancy mode is disabled When the active proxy doesn t respond to INVITE messages sent by the device the proxy is tagged as offline The behavior is similar to a Keep Alive OPTIONS or REGISTER failure Defines the Proxy keep alive time interval in seconds between Keep Alive messages This parameter is configured per Proxy Set The valid range is 5 to 2 000 000 The default value is 60 Note This parameter is applicable only if the parameter EnableProxyKeepAlive is set to 1 OPTIONS When the parameter EnableProxyKeepAlive is set to 2 REGISTER the time interval between Keep Alive messages is determined by the parameter RegistrationTime Enables the Proxy Hot Swap redundancy mode per Proxy Set 0 No Disabled default 1 Yes Proxy Hot Swap mode is enabled If Proxy Hot Swap is enabled the SIP INVITE REGISTER message is initially sent to the first Proxy Registrar server If there is no response from the first Proxy Registrar server after a specific number of retransmissions configured by the parameter HotSwapRtx the INVITE REGISTER message is resent to the next red
282. ber manipulation rules To use this feature with FXO interfaces configure the device to operate in one stage dialing mode lf this parameter is enabled it is possible to configure the FXS FXO interface to wait for dial tone per destination phone number before or during dialing of destination phone number Therefore the parameter IsWaitForDialTone configurable for the entire device is ignored The FXS interface send SIP 200 OK responses only after the DTMF dialing is complete The RFC 2833 DTMF relay dynamic payload type The valid range is 96 to 99 and 106 to 127 The default is 96 The 100 102 to 105 range is allocated for proprietary usage Notes Certain vendors e g Cisco use payload type 101 for RFC 2833 When RFC 2833 payload type negotiation is used i e the parameter TxDTMFOption is set to 4 this payload type is used for the received DTMF packets If negotiation isn t used this payload type is used for receive and for transmit 282 Document LTRT 65413 SIP User s Manual Parameter ReplaceNumberSignWithEs capeChar Web Special Digit Representation EMS Use Digit For Special DTMF UseDigitForSpecialDTMF 6 Configuration Parameters Reference Description Determines whether to replace the number sign with the escape character 23 in outgoing SIP messages for Tel to IP calls 0 Disable default 1 Enable All number signs received in the dialed DTMF digits ar
283. c 30 Third Burst Ring On Time 10msec 30 Third Burst Ring Off Time 10msec 30 Fourth Ring On Time 10msec 100 Fourth Ring Off Time 10msec 300 An example of various ringing signals definition is shown below NUMBER OF DISTINCTIVE RINGING PATTERNS Number of Ringing Patterns 3 Regular North American Ringing Pattern Ringing Pattern 0 Ring Type 0 Freq Hz 20 First Ring On Time 10msec 200 First Ring Off Time 10msec 400 GR 506 CORE Ringing Pattern 1 Ringing Pattern 1 Ring Type 1 Freq Hz 20 First Ring On Time 10msec 200 First Ring Off Time 10msec 400 GR 506 CORE Ringing Pattern 2 Ringing Pattern 2 Ring Type 2 Freq Hz 20 First Ring On Time 10msec 80 First Ring Off Time 10msec 40 Second Ring On Time 10msec 80 Second Ring Off Time 10msec 400 Version 6 0 371 March 2010 Aa K tal AudioCodes MediaPack Series 8 1 2 FXS Distinctive Ringing and Call Waiting Tones per Source Number The device supports the configuration of a Distinctive Ringing tone and Call Waiting Tone per calling number for IP to Tel calls This feature can be configured per FXS endpoint or for a range of FXS endpoints Therefore different tones can be played per FXS endpoint s depending on the source number of the received call This configuration is performed using the Tonelndex ini file table parameter which maps Ringing and or Call Waiting tones to source number prefixes per F
284. ce and highest bandwidth i e Full Duplex with 100Base TX but at the same time adhering to the guidelines listed above Note that when remote configuration is performed the device should be in the correct Ethernet setting prior to the time this parameter takes effect When for example the device is configured using BootP TFTP the device performs many Ethernet based transactions prior to reading the ini file containing this device configuration parameter To resolve this problem the device always uses the last Ethernet setup mode configured In this way if you want to configure the device to operate in a new network environment in which the current Ethernet setting of the device is invalid you should first modify this parameter in the current network so that the new setting holds next time the device is restarted After reconfiguration has completed connect the device to the new network and restart it As a result the remote configuration process that occurs in the new network uses a valid Ethernet configuration NAT Network Address Translation Support Network Address Translation NAT is a mechanism that maps a set of internal IP addresses used within a private network to global IP addresses providing transparent routing to end hosts The primary advantages of NAT include 1 Reduction in the number of global IP addresses required in a private network global IP addresses are only used to connect to the Internet 2 Better network se
285. ceMode InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway InterfaceTable VlanID InterfaceTable InterfaceName InterfaceTable 0 0 10 192 168 85 14 16 0 0 0 0 1 ManagementIF InterfaceTable 1 LO OO ZOOS Sel 222920 OP On On O 20 OY SOE OM A InterfaceTable 2 iL Ad ZZ 195 z al 247 ala aval 8S Ab 211 myMedialF InterfaceTable VLAN related parameters VlanMode 1 VlanNativeVlanId 1 Routing Table Configuration RoutingTableDestinationsColumn 176 85 49 0 RoutingTableDestinationPrefixLensColumn 24 RoutingTableGatewaysColumn 192 168 0 1 RoutingTableInterfacesColumn 0 RoutingTableHopsCountColumn 1 ou il N SIP User s Manual 462 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities Example 3 One interface exclusively for management OAMP applications and two others for Call Control and RTP CONTROL and MEDIA applications The Multiple Interface table is configured with four interfaces One is exclusively for Management and the two are for Call Control and RTP Media applications Two of them are IPv4 interfaces Table 10 13 Multiple Interface Table Example 3 Prefix Default VLAN Interface Index Application Interface IP Address Length Gateway ID Name 0 OAMP IPv4 192 168 85 14 16 0 0 0 0 1 Mgmt 1 Media IPv4 200 200 85 14 24 200 200 85 1 201 CntriMediat Control 2 Media amp IPv4 200 200 86 14 24 0 0 0 0 202 CntriMedia2 Control VLANs are
286. ceived from IP the Tel Display Name remains empty default 1 Yes If an IP Display Name is received it is used as the Tel Source Number and also as the Tel Display Name and Presentation is set to Allowed 0 If no Display Name is received from IP the IP Source Number is used as the Tel Source Number and Presentation is set to Restricted 1 For example When From 100 lt sip 200 201 202 203 204 gt is received the outgoing Source Number and Display Name are set to 100 and the Presentation is set to Allowed 0 When From lt sip 100 101 102 103 104 gt is received the outgoing Source Number is set to 100 and the Presentation is set to Restricted 1 Determines whether to use the device s routing table to obtain the URI host name and optionally an IP profile per call even if a Proxy server is used 0 Disable Don t use internal routing table default 1 Enable Use the Tel to IP Routing Notes This parameter appears only if the Use Default Proxy parameter is enabled The domain name is used instead of a Proxy name or IP address in the INVITE SIP URI For a description of this parameter refer to Configuring the Tel to IP Routing on page 126 This ini file table parameter configures the Tel to IP Routing for routing Tel to IP calls The format of this parameter is as follows PREFIX FORMAT PREFIX_Index PREFIX_DestinationPrefix P EFIX DestAddress PREFIX_SourcePrefix
287. clude this parameter using the BootP TFTP Server utility refer to the Product Reference Manual Resets the username and password of the primary and secondary accounts to their defaults 0 Password and username retain their values default 1 Password and username are reset for the default username and password refer to User Accounts Notes For this parameter to take effect a device reset is required The username and password cannot be reset from the Web interface i e via AdminPage or by loading an ini file Defines the file name of the Scenario file to be loaded to the device The file name must have the dat extension and can be up to 47 characters For loading a Scenario using the Web interface refer to Loading a Scenario to the Device on page 42 223 March 2010 ca AudioCodes Parameter WelcomeMessage MediaPack Series Description This ini file table parameter configures the Welcome message that appears after a Web interface login The format of this parameter is as follows WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_Text WelcomeMessage For Example WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_Text WelcomeMessage 1 WHEKKEKKRKEKEKREKKRERERERERERERERERERERM WelcomeMessage 2 This is a Welcome message WelcomeMessage 3 nkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkki WelcomeMessage Notes 6 2 3 Each index repres
288. cludes the following icons U U C m Globe 3 highest level in the tree from which a Region can be added m Region 4 defines a group e g geographical location to which devices can be added If you click a Region that is defined with devices MG s the Main pane see figure above displays a list of all the devices pertaining to the Region m MG defines the device This is the lowest level in the tree If you click an MG icon the Main pane see figure above displays a graphical representation of the device s chassis 191 March 2010 A e AudioCodes MediaPack Series 5 2 Securing EMS Device Communication 5 2 1 Configuring IPSec Before you can configure the device through the EMS you need to configure the secure communication protocol IPSec for communicating between the EMS and the device Before you enable IPSec in the EMS you must define the IPSec IKE pre shared key in a secure manner This is performed through an SSH secure shell client session e g PUTTY Once you have defined the IPSec IKE pre shared key you must enter the same IPSec IKE pre shared key in the EMS when you define the device Before performing the procedure below ensure that you have the following information m The IP address of the EMS Server that is to communicate with the device m An initial password for the IKE pre shared key The device is shipped with SSH enabled The configuration text is case and space sensitive
289. contains tgrp lt source trunk group ID gt trunk context lt gateway IP address gt The lt source trunk group ID gt is the Hunt Group ID where incoming calls from Tel is received For IP Tel calls the SIP 200 OK device s response contains tgrp lt destination trunk group ID gt trunk context lt gateway IP address gt The lt destination trunk group ID gt is the Hunt Group ID used for outgoing Tel calls The lt gateway IP address gt in trunk context can be configured using the parameter SIPGatewayName Note IP to Tel configuration using the parameter PSTNPrefix overrides the tgrp parameter in incoming INVITE messages Determines the precedence method for routing IP to Tel calls according to the IP to Hunt Group Routing Table or according to the SIP tgrp parameter 0 default IP to Tel routing is determined by the IP to Hunt Group Routing Table PSTNPrefix parameter If a matching rule is not found in this table the device uses the Hunt Group parameters for routing the call 1 The device first places precedence on the tgrp parameter for IP to Tel routing If the received INVITE Request URI does not contain the tgrp parameter or if the Hunt Group number is not defined then the IP to Hunt Group Routing Table is used for routing the call Below is an example of an INVITE Request URI with the tgrp parameter indicating that the IP call should be routed to Hunt Group 7 INVITE
290. criterion independently For available notations representing multiple numbers digits for destination and source prefixes refer to Dialing Plan Notation for Routing and Manipulation on page 377 For configuring number manipulation using ini file table parameters NumberMapIP2Tel NumberMapTel2IP SourceNumberMapIP2Tel and SourceNumberMapTel2IP refer to Number Manipulation and Routing Parameters on page 331 gt To configure the Number Manipulation tables 1 Open the required Number Manipulation page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt Dest Number IP gt Tel Dest Number Tel gt IP Source Number IP gt Tel or Source Number Tel gt IP page item the relevant Manipulation table page is displayed e g Source Phone Number Manipulation Table for Tel gt IP Calls page Figure 3 74 Source Phone Number Manipulation Table for Tel to IP Calls Index Source Trunk Group Source IP Group Destination Prefix Source Prefix Stripped Digits From Left Stripped Digits From Right l Leave Number of Digits to Prefix to Add Suffix to Add Presentation The figure above shows an example of the use of manipulation rules for Tel to IP source phone number manipulation e Index 1 When the destination number has the prefix 03 e g 035000 source number prefix 201 e g 20155 and from source IP Group ID 2 the source number is changed to for e
291. ctor Enable VoiceMail URI Retry After Time Enable P Associated URI Header Source Number Preference Forking Handling Mode Enable Comfort Tone Add Trunk Group ID as Prefix to Source Enable Reason Header 0 0 0 0 Supported By Dest Phone Number Disable Progress 0 30 Re NVITE Disabled No Fax Initiate T 38 on Preamble UDP 5060 5060 5061 Disable Enable 0 5060 Yes No Disable 180 Disable Yes Disable No No Disable Don t Play Play According to Early Media Disable Disable IEA AudiocodesGw None Disable Forward Disable Disable 0 Disable Parallel handling Disable No Enable Retransmission Parameters SIP T1 Retransmission Timer msec SIP T2 Retransmission Timer msec SIP Maximum RTX Configure the parameters as required Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 89 March 2010 7a K tal AudioCodes MediaPack Series 3 3 4 3 2 Configuring DTMF and Dialing Parameters The DTMF 8 Dialing page is used to configure paramete
292. curity by hiding its internal architecture Version 6 0 443 March 2010 7a K tal AudioCodes MediaPack Series The following figure illustrates the device s supported NAT architecture Figure 10 1 Nat Functioning MediaPack m The design of SIP creates a problem for VolP traffic to pass through NAT SIP uses IP addresses and port numbers in its message body and the NAT server can t modify SIP messages and therefore can t change local to global addresses Two different streams traverse through NAT signaling and media A device located behind a NAT that initiates a signaling path has problems in receiving incoming signaling responses they are blocked by the NAT server Furthermore the initiating device must notify the receiving device where to send the media To resolve these issues the following mechanisms are available m STUN refer to STUN on page 444 m First Incoming Packet Mechanism refer to First Incoming Packet Mechanism on page 445 m RTP No Op packets according to the avt rtp noop draft refer to No Op Packets on page 446 For information on SNMP NAT traversal refer to the Product Reference Manual 10 2 1 STUN Simple Traversal of UDP through NATs STUN based on RFC 3489 is a client server protocol that solves most of the NAT traversal problems The STUN server operates in the public Internet and the STUN clients are embedded in end devices located behind NAT STUN is used both for the signal
293. d according to the following scenarios v During an existing call if the user presses Flash the call is put on hold a dial tone is heard and the user is able to initiate a second call Once the second call is established on hooking transfers the first held call to the second call v During an existing call if a call comes in call waiting pressing Flash places the active call on hold and answers the waiting call pressing Flash again toggles between these two calls 1 1 Sequence of Flash hook digit v Flash 1 holds a call or toggles between two existing calls v Flash 2 makes a call transfer v Flash 3 makes a three way conference call if the Three Way Conference feature is enabled i e the parameter Enable3WayConference is set to 1 and the parameter 3WayConferenceMode is set to 2 Flash keys sequence timeout the time in msec that the device waits for digits after the user presses the Flash button Flash Hook Digit mode when the parameter FlashKeysSequenceStyle is set to 1 The valid range is 100 to 5 000 The default is 2 000 Keypad Feature Call Forward Parameters Web Unconditional EMS Call Forward Unconditional KeyCFUnCond Web No Answer EMS Call Forward No Answer KeyCFNoAnswer Web On Busy EMS Call Forward Busy KeyCFBusy Web On Busy or No Answer EMS CF Busy Or No Answer KeyCFBusyOrNoAnswer Web Do Not Disturb EMS CF Do Not Disturb KeyCFDoNotDisturb Keypa
294. d Telnet Parameters Description Defines up to ten IP addresses that are permitted to access the device s Web interface and Telnet interfaces Access from an undefined IP address is denied When no IP addresses are defined in this table this security feature is inactive i e the device can be accessed from any IP address The default value is 0 0 0 0 i e the device can be accessed from any IP address For example WebAccessList 0 10 13 2 66 WebAccessList 1 10 13 77 7 For defining the Web and Telnet Access list using the Web interface refer to Configuring the Web and Telnet Access List on page 69 Uses RADIUS queries for Web and Telnet interface authentication 0 Disable default 1 Enable When enabled logging in to the device s Web and Telnet embedded servers is performed through a RADIUS server The device contacts a user defined server and verifies the given user name and password pair against a remote database in a secure manner Notes The parameter EnableRADIUS must be set to 1 RADIUS authentication requires HTTP basic authentication meaning the user name and password are transmitted in clear text over the network Therefore it s recommended to set the parameter HTTPSOnly to 1 to force the use of HTTPS since the transport is encrypted fusing RADIUS authentication when logging in to the CLI only the primary Web User Account which has Security Administration access level can access
295. d calls The ACD value is refreshed every 15 minutes and therefore this value reflects the average duration of all established calls made within a 15 minute period Indicates the number of attempted fax calls Indicates the number of successful fax calls 3 5 2 2 Viewing SAS Registered Users The SAS Registered Users page displays a list of registered users gt To view the registered users m Open the SAS Registered Users page Status amp Diagnostics tab gt Gateway Statistics menu gt SAS Registered Users page item Address Of Record lt sip 2400 Proxies ac gt lt sip 2401 Proxies ac gt lt sip 2500 Proxies ac gt lt sip 2402 Proxies ac gt lt sip 2403 Proxies ac gt lt sip 2404 Proxies ac gt lt sip 2405 Proxies ac gt Column Name Address of Record Figure 3 115 SAS Registered Users Page Contact lt sip 2400 10 8 210 5 gt expires 160 lt sip 2401 10 8 210 5 gt expires 160 lt sip 2500 10 8 210 5 gt expires 180 lt sip 2402 10 6 210 5 gt expires 160 lt sip 2403 10 8 210 5 gt expires 160 lt sip 2404 10 6 210 5 gt expires 160 lt sip 2405 10 8 210 5 gt expires 160 Table 3 34 SAS Registered Users Parameters Description An address of record AOR is a SIP or SIPS URI that points to a domain with a location service that can map the URI to another URI Contact where the user might be available Contact SIP URI that can be used to contact that specific instance of the
296. d relates to the overall re assembled packet size and not to the size of each fragment Expected traffic rate bytes per second Tolerance of traffic rate limit number of bytes Action upon match i e Allow or Block A read only field providing the number of packets accepted rejected by the specific rule 72 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 3 4 3 3 3 4 1 Version 6 0 Configuring the Certificates The Certificates page is used for both HTTPS and SIP TLS secure communication Replacing the server certificate refer to Server Certificate Replacement on page 73 Replacing the client certificates refer to Client Certificates on page 75 Regenerating Self Signed Certificates refer to Self Signed Certificates on page 76 Updating the private key using HTTPSPkeyFileName as described in the Product Reference Manual Server Certificate Replacement The device is supplied with a working Secure Socket Layer SSL configuration consisting of a unique self signed server certificate If an organizational Public Key Infrastructure PKI is used you may wish to replace this certificate with one provided by your security administrator gt 1 To replace the device s self signed certificate Your network administrator should allocate a unique DNS name for the device e g dns_name corp customer com This DNS name is used to access the device and should therefore
297. d sequence that activates the immediate call forward option Keypad sequence that activates the forward on no answer option Keypad sequence that activates the forward on busy option Keypad sequence that activates the forward on busy or no answer option Keypad sequence that activates the Do Not Disturb option immediately reject incoming calls To activate the required forward method from the telephone 1 Dial the user defined sequence number on the keypad a dial tone is heard 2 Dial the telephone number to which the call is forwarded terminate the number with a confirmation tone is heard Version 6 0 325 March 2010 ca AudioCodes Parameter Web Deactivate EMS Call Forward Deactivation KeyCFDeact MediaPack Series Description Keypad seguence that deactivates any of the call forward options After the seguence is pressed a confirmation tone is heard Keypad Feature Caller ID Restriction Parameters Web Activate EMS CLIR KeyCLIR Web Deactivate EMS CLIR Deactivation KeyCLIRDeact Keypad sequence that activates the restricted Caller ID option After the sequence is pressed a confirmation tone is heard Keypad sequence that deactivates the restricted Caller ID option After the sequence is pressed a confirmation tone is heard Keypad Feature Hotline Parameters Web Activate EMS Hot Line KeyHotLine Web Deactivate EMS Hot Line Deactivation KeyHotLineDeact Ke
298. d then off hook again within the user defined regret timeout configured by the parameter RegretTime Therefore the device notifies the far end that the call has been re answered 0 Disable default 1 Enable This parameter is typically implemented for incoming IP to Tel collect calls to the FXS port If the FXS user does not wish to accept the collect call the user disconnects the call by on hooking the phone The device notifies the softswitch or Application server of the unanswered collect call on hook by sending a SIP INFO message As a result the softswitch disconnects the call sends a BYE message to the device If the call is a regular incoming call and the FXS user on hooks the phone without intending to disconnect the call the softswitch does not disconnect the call during the regret time The INFO message format is as follows INFO sip 12345 10 50 228 164 5082 SIP 2 0 Via SIP 2 0 UDP 127 0 0 1 branch z9hG4bK_05_905924040 90579 From lt sip 551137077803 ims acme com br 5080 user phone gt t ag 008277765 To lt sip notavailable unknown invalid gt tag svw 0 1229428367 Call ID ConorCCR 0 LU 1229417827103300 dtas stdn fs5000group0 000 1 CSeq 1 INFO Contact sip 10 20 7 70 5060 Content Type application On Hook application Off Hook Content Length 0 Notes This parameter is applicable only if the parameter RegretTime is configured This parameter is applicable only to FXS interfaces
299. decibels This parameter sets the level for the received Tel to IP signal The valid range is 32 to 31 dB The default value is 0 dB Voice gain control in decibels This parameter sets the level for the transmitted IP to Tel signal The valid range is 32 to 31 dB The default value is 0 dB Determines the bit ordering of the G 726 G 727 voice payload format 0 Little Endian default 1 Big Endian Note To ensure high voice quality when using G 726 G 727 both communicating ends should use the same endianness format Therefore when the device communicates with a third party entity that uses the G 726 G 727 voice coder and voice quality is poor change the settings of this parameter between Big Endian and Little Endian Currently not supported Currently not supported Determines in 100 msec resolution the time between activating the Answer Detector and the time that the detector actually starts to operate The valid range is 0 to 1023 The default is 0 Currently not supported Currently not supported Determines the Answer Detector sensitivity The range is 0 most sensitive to 2 least sensitive The default is 0 Silence Suppression is a method for conserving bandwidth on VoIP calls by not sending packets when silence is detected 0 Disable Silence Suppression is disabled default 1 Enable Silence Suppression is enabled 349 March 2010 ca AudioCodes Parameter
300. des MediaPack Series 6 10 Answer and Disconnect Supervision Parameters The answer and disconnect supervision parameters are described in the table below Table 6 47 Answer and Disconnect Parameters Parameter Web Answer Supervision EMS Enable Voice Detection EnableVoiceDetection Web EMS Max Call Duration min MaxCallDuration Web EMS Disconnect on Dial Tone DisconnectOnDialTone Web Send Digit Pattern on Connect EMS Connect Code TelConnectCode Web Disconnect on Broken Connection EMS Disconnect Calls on Broken Connection DisconnectOnBrokenConnection SIP User s Manual Description Enables the sending of SIP 200 OK upon detection of speech fax or modem 1 Yes The device sends SIP 200 OK to INVITE messages when speech fax modem is detected 0 No The device sends SIP 200 OK only after it completes dialing default Typically this feature is used only when early media EnableEarlyMedia is used to establish the voice path before the call is answered Notes This feature is applicable only to one stage dialing FXO This parameter is applicable only to FXO interfaces Defines the maximum call duration in minutes If this time expires both sides of the call are released IP and Tel The valid range is 0 to 35 791 The default is 0 i e no limitation Determines whether the device disconnects a call when a dial tone is detected from the PBX 0 Disable Cal
301. description refer to the device s nstallation Manual SIP User s Manual 366 Document LTRT 65413 SIP User s Manual 8 Auxiliary Configuration Files 8 8 1 Auxiliary Configuration Files This section describes the auxiliary files that can be loaded in addition to the ini file to the device m Call Progress Tones refer to Call Progress Tones File on page 367 m Distinctive Ringing in the ini file refer to Distinctive Ringing on page 370 m Prerecorded Tones refer to Prerecorded Tones File on page 372 m Dial Plan refer to Dial Plan File on page 373 m User Information refer to User Information File on page 374 You can load these auxiliary files to the device using one of the following methods m Loading the files directly to the device using the device s Web interface refer to Loading Auxiliary Files on page 163 m Specifying the auxiliary file name in the ini file refer to Auxiliary and Configuration Files Parameters on page 361 and then loading the ini file to the device Call Progress Tones File The Call Progress Tones CPT and Distinctive Ringing auxiliary file is comprised of two sections m The first section contains the definitions of the Call Progress Tones levels and frequencies that are detected generated by the device m The second section contains the characteristics of the Distinctive Ringing signals that are generated by the device refer to Distinctive Ringing on page 370 You can us
302. dication The device supports Message Waiting Indication MWI according to IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to MWI server The FXS device can accept an MWI NOTIFY message that indicates waiting messages or that the MWI is cleared Users are informed of these messages by a stutter dial tone The stutter and confirmation tones are defined in the CPT file refer to the Product Reference Manual If the MWI display is configured the number of waiting messages is also displayed If the MWI lamp is configured the phone s lamp on a phone that is equipped with an MWI lamp is lit The device can subscribe to the MWI server per port usually used on FXS or per device used on FXO To configure MWI use the following parameters EnableMWI MWIServerlP MWIAnalogLamp MWiIDisplay StutterToneDuration EnableMWISubscription MWIExpirationTime SubscribeRetryTime SubscriptionMode CallerlDType determines the standard for detection of MWI signals ETSIVMWITypeOneStandard BellcoreVMWITypeOneStandard Caller ID This section discusses the device s Caller ID support Caller ID Detection Generation on the Tel Side By default generation and detection of Caller ID to the Tel side is disabled To enable Caller ID set the parameter EnableCallerID to 1 When the Caller ID service is enabled m For FXS the Caller ID signal is sent to the device s port m For FXO the Caller ID signal is detected SIP User s Manua
303. dvanced Parameters Some pages provide you with an Advanced Parameter List Basic Parameter List toggle button that allows you to show or hide advanced parameters in addition to displaying the basic parameters This button is located on the top right corner of the page and has two states m Advanced Parameter List button with down pointing arrow click this button to display all parameters m Basic Parameter List button with up pointing arrow click this button to show only common basic parameters The figure below shows an example of a page displaying basic parameters only and then showing advanced parameters as well using the Advanced Parameter List button Figure 3 7 Toggling between Basic and Advanced Page View Declare RFC 2833 in SOP No ist Tx OTMF Option RFC 2833 2nd Tx OTMF Oplbon Jed Tx DTMF Option 4th Tx DTMF Option Sth Tx DTMF Option RFC 2032 Payload Type Ovefauk Destinabon Number Spean Dipt Represertamon Max Dvg s In Phone Num 5 Inter Digt Timesut fer Overlap Crating 4 sec Dedare RFC 2633 m SOP No ist Tx OTMF Option RFC Zed Tx DTMF Option Sed Tx DTMF Option 4th Tx DTMF Option Sth Tx DTMF Option RFC 2833 Payload Type Mook Flash Opbsn Opt Mapping Rules Dual Tone Duration sec Hothne Dial Tone Durabon sec Enable Specal Dozst Defaut Destinabon Number Specal Diot Regresertabon For ease of identification the basic parameters are displayed with a darker blue color backg
304. e page Configuration tab gt Security Settings menu gt IPSec Proposal Table Figure 3 54 IP Security Proposals Table Index Encryption Algorithm Authentication Algorithm Diffie Hellman Group 0 Oj 1 O In the figure above two proposals are defined e Proposal 0 AES SHA1 DH group 2 e Proposal 1 3DES SHA1 DH group 2 Note that with this configuration neither DES nor MD5 can be negotiated 2 Select an Index click Edit and then modify the proposal as required 3 Click Apply 4 To save the changes to flash memory refer to Saving Configuration on page 161 To delete a proposal select the relevant Index number and then click Delete Table 3 10 IP Security Proposals Table Configuration Parameters Parameter Name Description Encryption Algorithm Determines the encryption privacy algorithm IPsecProposalTable EncryptionAlgorithm 0 NONE 1 DES CBC 2 3DES CBC 3 AES default Authentication Algorithm Determines the message authentication IPsecProposalTable_AuthenticationAlgorithm integrity algorithm 0 NONE 2 HMAC SHAT1 96 4 HMAC MD5 96 default Version 6 0 79 March 2010 7a c tal AudioCodes MediaPack Series Parameter Name Description Diffie Hellman Group Determines the length of the key created by the IPsecProposalTable_DHGroup DH protocol for up to four proposals For the ini file parameter X depicts the proposal number 0
305. e InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway Interface Table VlanlD InterfaceTable InterfaceName Mnterface Table For example Interface Table 0 0 0 192 168 85 14 16 0 0 0 0 1 Management Interface Table 1 2 0 200 200 85 14 24 0 0 0 0 200 Control Interface Table 2 1 0 211 211 85 14 24 211 211 85 1 211 Media The above example configures three network interfaces OAMP Control and Media Notes For this ini file table parameter to take effect a device reset is reguired Up to 16 logical IP addresses with associated VLANs can be defined indices 0 15 However only up to 8 interfaces SIP User s Manual 208 Document LTRT 65413 SIP User s Manual Parameter Single IP Network Parameters Web IP Address EMS Local IP Address LocalOAMIPAddress Web Subnet Mask EMS OAM Subnet Mask LocalOAMSubnetMask Web Default Gateway Address EMS Local Def GW LocalOAMDefaultGW Version 6 0 6 Configuration Parameters Reference Description can be used for media RTP traffic assigned to a Media Realm in the SIP Media Realm table which in turn is assigned to an IP Group Each interface index must be unique Each IP interface must have a unique subnet Subnets in different interfaces must not be overlapping in any way e g defining two interfaces with 10 0 0 1 8 and 10 50 10 1 24 is invalid Each interface must have its own address
306. e configured Proxies the TLS connection is established The comparison is performed if the SubjectAltName is either a DNS name DNSName or an IP address If no match is found and the SubjectAltName is marked as critical the TLS connection is not established If DNSName is used the certificate can also use wildcards to replace parts of the domain name 236 Document LTRT 65413 SIP User s Manual Parameter 6 Configuration Parameters Reference Description If the SubjectAltName is not marked as critical and there is no match the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName If a match is found the connection is established Otherwise the connection is terminated Web TLS Client Verify Server Determines whether the device when acting as a client for TLS Certificate connections verifies the Server certificate The certificate is EMS Verify Server Certificate verified with the Root CA information VerifyServerCertificate 0 Disable default 1 Enable Note If Subject Name verification is necessary the parameter PeerHostNameVerificationMode must be used as well Web EMS TLS Remote Subject Defines the Subject Name that is compared with the name Name TLSRemoteSubjectName defined in the remote side certificate when establishing TLS connections If the SubjectAltName of the received certificate is not equal to any of the defined
307. e Dialing One stage dialing is when the FXO device receives an IP to Tel call off hooks the PBX line connected to the telephone and then immediately dials the destination telephone number In other words the IP caller doesn t dial the PSTN number upon hearing a dial tone Figure 9 4 Call Flow for One Stage Dialing FXO Gateway SIP Client F1 INVITE FXO seizes line FXO waits for dial tone from PBX if defined by IsvVaitForDialTone and VVaitF orDialTone F4 200 OK immediatley or after detecting polarity reversal or voice One stage dialing incorporates the following FXO functionality m Waiting for Dial Tone Enables the device to dial the digits to the Tel side only after detecting a dial tone from the PBX line The ini file parameter IsWaitForDialTone is used to configure this operation m Time to Wait Before Dialing Defines the time in msec between seizing the FXO line and starting to dial the digits The ini file parameter WaitForDialTime is used to configure this operation Note The ini file parameter IsWaitForDialTone must be disabled for this mode Version 6 0 387 March 2010 7a K tal AudioCodes MediaPack Series m Answer Supervision The Answer Supervision feature enables the FXO device to determine when a call is connected by using one of the following methods e Polarity Reversal device sends a 200 OK in response to an INVITE only when it detects a polarity reversal e Voice Detection device
308. e PBX to indicate call forward with no reason when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on busy when the original call is received from an external line not an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on no answer when the original call is received from an external line not an internal extension The valid range is a 120 character string 276 Document LTRT 65413 SIP User s Manual Parameter Web Forward on Do Not Disturb Digit Pattern External EMS VM Digit Pattern On DND External DigitPatternForwardOnDNDExt Web Forward on No Reason Digit Pattern External EMS VM Digit Pattern No Reason External DigitPatternForwardNoReasonExt Web Internal Call Digit Pattern EMS Digit Pattern Internal Call DigitPatternInternalCall Web External Call Digit Pattern EMS Digit Pattern External Call DigitPatternExternalCall Web Disconnect Call Digit Pattern EMS Tel Disconnect Code TelDisconnectCode Web Digit To Ignore Digit Pattern EMS Digit To Ignore DigitPatternDigitTolgnore 6 7 4 6 Configuration Parameters Reference Description Determines the digit pattern used by the PBX to indicate call forward on do not disturb when the original call is received from
309. e Source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 To configure manipulation of source numbers for IP to Tel calls using the Web interface refer to Configuring the Number Manipulation Tables on page 115 Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 SIP User s Manual 348 Document LTRT 65413 SIP User s Manual 6 16 6 Configuration Parameters Reference Channel Parameters This subsection describes the device s channel parameters 6 16 1 Voice Parameters The voice parameters are described in the table below Table 6 57 Voice Parameters Parameter Web EMS Input Gain InputGain Web Voice Volume EMS Volume dB VoiceVolume EMS Payload Format VoicePayloadFormat Web MF Transport Type MFTransportType Web Enable Answer Detector EnableAnswerDetector Web Answer Detector Activity Delay AnswerDetectorActivityDelay Web Answer Detector Silence Time AnswerDetectorSilenceTime Web Answer Detector Redirection AnswerDetectorRedirection Web Answer Detector Sensitivity EMS Sensitivity AnswerDetectorSensitivity Web Silence Suppression EMS Silence Compression Mode EnableSilenceCompression Version 6 0 Description Pulse code modulation PCM input gain control in
310. e a new scenario Note If a Scenario already exists the Scenario Loading message box appears 2 Click OK the Scenario mode appears in the Navigation tree as well as the menus of the Configuration tab Note If a Scenario already exists and you wish to create a new one click the Create Scenario button and then click OK in the subsequent message box 3 Inthe Scenario Name field enter an arbitrary name for the Scenario 4 On the Navigation bar click the Configuration or Management tab to display their respective menus in the Navigation tree 5 Inthe Navigation tree select the required page item for the Step and then in the page itself select the required parameters by selecting the check boxes corresponding to the parameters 6 Inthe Step Name field enter a name for the Step Version 6 0 af March 2010 ca AudioCodes 7 Click the Next button located at the bottom of the page the Step is added to the Scenario and appears in the Scenario Step list entouaton Manepenert pramostes Scenarios Search Basic Full Network Settings Maca Setbags Secunty Setungs gt i Protocol Configuraton ud Protocol Oefrstson SIP General Parameters Proxy Registrabon Coders DTMF amp Dialing PUBSIP Advanced Parameters B Manipulation Tables Selected Page Mook Flash Option Not Suppoted Figure 3 15 Creating a Scenario Selected Parameter Deck Porameler st a hd Max Dagits In Pho
311. e button corresponding to the file type that you want to load navigate to the folder in which the file is located and then click Open the name and path of the file appear in the field next to the Browse button 3 Click the Load File button corresponding to the file you want to load 4 Repeat steps 2 through 3 for each file you want to load 5 To save the loaded auxiliary files to flash memory refer to Saving Configuration on page 161 6 To reset the device if you have loaded a Call Progress Tones file refer to Resetting the Device on page 159 You can also load the auxiliary files using the ini file loaded to the device using BootP Each auxiliary file has a specific ini file parameter that specifies the name of the auxiliary file that you want to load to the device For a description of these ini file parameters refer to Configuration Files Parameters on page 361 SIP User s Manual 164 Document LTRT 65413 SIP User s 3 4 2 2 Version 6 0 Manual 3 Web Based Management To load the auxiliary files using an ini file 1 In the ini file define the auxiliary files to be loaded to the device You can also define in the ini file whether the loaded files must be stored in the non volatile memory so that the TFTP process is not required every time the device boots up 2 Save the auxiliary files and the ini file in the same directory on your local PC 3 Invoke a BootP TFTP session the ini and associated auxilia
312. e configured by the Minimum delay parameter For certain scenarios the Optimization Factor is set to 13 One of the purposes of the Jitter Buffer mechanism is to compensate for clock drift If the two sides of the VoIP call are not synchronized to the same clock source one RTP source generates packets at a lower rate causing under runs at the remote Jitter Buffer In normal operation optimization factor O to 12 the Jitter Buffer mechanism detects and compensates for the clock drift by occasionally dropping a voice packet or by adding a BFI packet Fax and modem devices are sensitive to small packet losses or to added BFI packets Therefore to achieve better performance during modem and fax calls the Optimization Factor should be set to 13 In this special mode the clock drift correction is performed less frequently only when the Jitter Buffer is completely empty or completely full When such condition occurs the correction is performed by dropping several voice packets simultaneously or by adding several BFI packets simultaneously so that the Jitter Buffer returns to its normal condition Version 6 0 441 March 2010 A c tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 442 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities 10 10 1 10 2 Networking Capabilities This section provides an overview of the device s networking capabilities Ethernet Interface Configuration The device
313. e default is an empty field Selects the Proxy Set ID defined in Configuring the Proxy Sets Table on page 97 to associate with the IP Group All INVITE messages configured to be sent to the specific IP Group are in fact sent to the IP address associated with this Proxy Set The range is 1 5 Note Proxy Set ID O must not be selected this is the device s default Proxy The request URI host name used in INVITE and REGISTER messages that are sent to this IP Group or the host name in the From header of INVITE messages received from this IP Group If not specified the value of the global parameter ProxyName refer to Configuring the Proxy and Registration Parameters on page 96 is used instead The value range is a string of up to 49 characters The default is an empty field Defines the user part for the From To and Contact headers of SIP REGISTER messages and the user part for the Contact header of INVITE messages that are received from this IP Group and forwarded by the device to another IP Group Note This parameter is overridden by the Contact User parameter in the Account table refer to Configuring the Account Table on page 93 The IP Profile that you want assigned to this IP Group The default is 0 Note IP Profiles are configured using the parameter IPProfile refer to Configuring P Profile Settings on page 107 Determines the Request URI host name in outgoing INVITE messages 0 Disable
314. e default value is 0 i e RTP multiplexing is disabled Notes For this parameter to take effect a device reset is reguired All devices that participate in the same RTP multiplexing session must use this same port Enables or disables the transmission of RTP or T 38 No Op packets 0 Disable default 1 Enable This mechanism ensures that the NAT binding remains open during RTP or T 38 silence periods Defines the time interval in which RTP or T 38 No Op packets are sent in the case of silence no RTP T 38 traffic when No Op packet transmission is enabled The valid range is 20 to 65 000 msec The default is 10 000 Note To enable No Op packet transmission use the NoOpEnable parameter Determines the payload type of No Op packets The valid range is 96 to 127 for the range of Dynamic RTP Payload Type for all types of non hard coded RTP Payload types refer to RFC 3551 The default value is 120 Note When defining this parameter ensure that it doesn t cause collision with other payload types Defines the time interval in msec between adjacent RTCP reports The interval range is 0 to 65 535 The default interval is 5 000 Controls whether RTCP report intervals are randomized or whether each report interval accords exactly to the parameter RTCPInterval 0 Disable Randomize default 1 Enable No Randomize 360 Document LTRT 65413 SIP User s Manual 6 Configuration Paramet
315. e erased and balancing starts over again When the Random Weights algorithm is used the outgoing requests are not distributed equally among the Proxies The weights are received from the DNS server by using SRV records The device sends the requests in such a fashion that each Proxy receives a percentage of the requests according to its assigned weight A single FQDN should be configured as a Proxy IP address The Random Weights Load Balancing is not used in the following scenarios The Proxy Set includes more than one Proxy IP address The only Proxy defined is an IP address and not an FQDN SRV is not enabled DNSQueryType The SRV response includes several records with a different Priority value Determines whether Keep Alive with the Proxy is enabled or disabled This parameter is configured per Proxy Set 0 Disable Disable default 1 Using OPTIONS Enables Keep Alive with Proxy using OPTIONS 2 Using REGISTER Enable Keep Alive with Proxy using REGISTER If set to Using OPTIONS the SIP OPTIONS message is sent every user defined interval as configured by the parameter ProxyKeepAliveTime If set to Using REGISTER the SIP REGISTER message is sent every user defined interval as configured by the parameter RegistrationTime Any response from the Proxy either success 200 OK or failure 4xx response is considered as if the Proxy is communicating correctly Notes This parameter must
316. e following options e Yes The device s current configuration is saved burned to the flash memory prior to reset default e _ No Resets the device without saving the current configuration to flash discards all unsaved modifications 159 March 2010 A c tal AudioCodes MediaPack Series 3 Under the Reset Configuration group from the Graceful Option drop down list select one of the following options e Yes Reset starts only after the user defined time in the Shutdown Timeout field refer to Step 4 expires or after no more active traffic exists the earliest thereof In addition no new traffic is accepted e No Reset starts regardless of traffic and any existing traffic is terminated at once 4 In the Shutdown Timeout field relevant only if the Graceful Option in the previous step is set to Yes enter the time after which the device resets Note that if no traffic exists and the time has not yet expired the device resets 5 Click the Reset button a confirmation message box appears requesting you to confirm Figure 3 102 Reset Confirmation Message Box r Microsoft Internet Explorer j Are you sure you want to RESET the Gateway 6 Click OK to confirm device reset if the parameter Graceful Option is set to Yes in Step 3 the reset is delayed and a screen displaying the number of remaining calls and time is displayed When the device begins to reset a message appea
317. e ini file is AudioCodes e The equation to be evaluated is according to RFC this part is called A1 122 audiocodes com AudioCodes e The MD5 algorithm is run on this equation and stored for future usage e The result is a8f17d4b41ab8dab6c95d3c14e34a9e1 5 Next the par called A2 needs to be evaluated e The method type is REGISTER e Using SIP protocol sip e Proxy IP from ini file is 10 2 2 222 e The equation to be evaluated is REGISTER sip 10 2 2 222 e The MD5 algorithm is run on this equation and stored for future usage e The result is a9a031cfddcb10d91c8e 7b4926086f7e SIP User s Manual 424 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 6 Final stage e The A1 result The nonce from the proxy response is 11432d6bce58ddf02e3b5e1c77c010d2 e The A2 result The equation to be evaluated is A1 11432d6bce58ddf02e3b5e1c77c010d2 A2 e The MD65 algorithm is run on this equation The outcome of the calculation is the response needed by the device to register with the Proxy e The response is b9c45d0234a5abfoddf5c704029b38cf At this time a new REGISTER request is issued with the following response REGISTERS pO 22222 SIP 2 0 Was SLP 2 0 UDE LO i1 1 200 From lt Sip 122 10 1 1 200 gt tag 1c23940 MOR S 220102 0055 Call ID 654982194 10 1 1 200 Server Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeq 1 REGISTER Con
318. e item Figure 3 49 Firewall Settings Page Edit Is Rule Prefix uses Port Burst Action Match Rule Active Source IP Length Range Protocol Packet Size Byte ratei Bytes ME an csin 192 0 0 0 8 0 65535 Any 40000 50000 ALLOW 0 10 31 4 0 24 4000 9000 Any 0 0 BLOCK 0 10 4 0 0 16 4000 9000 Any 0 0 BLOCK 0 SIP User s Manual 70 Document LTRT 65413 SIP User s Manual 3 Web Based Management 5 In the Add field enter the index of the access rule that you want to add and then click Add a new firewall rule index appears in the table Configure the firewall rule s parameters according to the table below Click one of the following buttons e Apply saves the new rule without activating it e Duplicate Rule adds a new rule by copying a selected rule e Activate saves the new rule and activates it e Delete deletes the selected rule To save the changes to flash memory refer to Saving Configuration on page 161 The figure above shows the following access list settings je S B OS v N a N N Version 6 0 Rule 1 traffic from the host mgmt customer com destined to TCP ports 0 to 80 is always allowed Rule 2 traffic from the 192 xxx yyy zzz subnet is limited to a rate of 40 Kbytes per second with an allowed burst of 50 Kbytes Note that the rate is specified in bytes not bits per second a rate of 40000 bytes per second nominally corresponds to 320 kbps
319. e of the User Information which is loaded to the device in the User Information auxiliary file For a description on User Information refer to Loading Auxiliary Files on page 163 0 Disable default 1 Enable 256 Document LTRT 65413 SIP User s Manual Parameter HandleReasonHeader EnableSilenceSuppInSDP EnableRport EMS X Channel Header XChannelHeader Version 6 0 6 Configuration Parameters Reference Description Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping 0 Disregard Reason header in incoming SIP messages 1 Use the Reason header value for Release Reason mapping default Determines the device s behavior upon receipt of SIP Re INVITE messages that include the SDP s silencesupp off attribute 0 Disregard the silecesupp attribute default 1 Handle incoming Re INVITE messages that include the silencesupp off attribute in the SDP as a request to switch to the Voice Band Data VBD mode In addition the device includes the attribute a silencesupp off in its SDP offer Note This parameter is applicable only if the G 711 coder is used Enables or disables the usage of the rport parameter in the Via header 0 Enabled 1 Disabled default The device adds an rpor parameter to the Via header of each outgoing SIP message The first Proxy that receives this message sets the
320. e one of the supplied auxiliary files dat file format or create your own file To create your own file it s recommended to modify the supplied usa_tone ini file in any standard text editor to suit your specific requirements and then convert the modified ini file into binary format using the TrunkPack Downloadable Conversion Utility DConvert For a description on converting a CPT ini file into a binary dat file refer to the Product Reference Manual Note Only the dat file format can be loaded to the device You can create up to 32 different Call Progress Tones each with frequency and format attributes The frequency attribute can be single or dual frequency in the range of 300 to 1980 Hz or an Amplitude Modulated AM Up to 64 different frequencies are supported Only eight AM tones in the range of 1 to 128 kHz can be configured the detection range is limited to 1 to 50 kHz Note that when a tone is composed of a single frequency the second frequency field must be set to zero The format attribute can be one of the following m Continuous A steady non interrupted sound e g a dial tone Only the First Signal On time should be specified All other on and off periods must be set to zero In this case the parameter specifies the detection period For example if it equals 300 the tone is detected after 3 seconds 300 x 10 msec The minimum detection time is 100 msec Version 6 0 367 March 2010 A tal AudioCodes
321. e only through the ini file and not the Web interface To restore the device to default settings using the ini file refer to 4 1 Restoring Factory Default Settings on page 365 INI File Format The ini file can be configured with any number of parameters These ini file parameters can be one of the following parameter types m Individual parameters refer to Configuring Individual ini File Parameters on page 185 m Table parameters refer to Configuring ini File Table Parameters on page 186 Configuring Individual ini File Parameters The format of individual ini file parameters includes an optional subsection name group name to conveniently group similar parameters by their functionality Following this line are the actual parameter settings These format lines are shown below subsection name the subsection name is optional Parameter Name Parameter Value Parameter Name Parameter Value Remark For general ini file formatting rules refer to General ini File Formatting Rules on page 188 Version 6 0 185 March 2010 Aa c tal AudioCodes MediaPack Series An example of an ini file containing individual ini file parameters is shown below System Parameters SyslogServerIP 10 13 2 69 EnableSyslog 1 these are a few of the system related parameters Web Parameters LogoWidth 339 WebLogoText My Device UseWeblogo 1 these are a few of the Web related pa
322. e opens the stream toward any subsequent 18x responses with an SDP Note Regardless of this parameter value once a SIP 200 OK response is received the device uses the RTP information and re opens the voice stream if necessary The timeout in seconds that is started after the first SIP 2xx response has been received for a User Agent when a Proxy server performs call forking Proxy server forwards the INVITE to multiple SIP User Agents The device sends a SIP ACK and BYE in response to any additional SIP 2xx received from the Proxy within this timeout Once this timeout elapses the device ignores any subsequent SIP 2xx The number of supported forking calls per channel is 4 In other words for an INVITE message the device can receive up to 4 forking responses from the Proxy server The valid range is 0 to 30 The default is 30 Enables or disables the usage of the SIP Reason header 0 Disable 1 Enable default Assigns a name to the device e g device123 com Ensure that the name you choose is the one with which the Proxy is configured to identify the device Note If specified the device name is used as the host part of the SIP URI in the From header If not specified the device s IP address is used instead default Determines the device s response to an incoming SDP that includes an IP address of 0 0 0 0 in the SDP s Connection Information field i e c IN IP4 0 0 0 0 0 Sets the IP address of
323. e replaced in the outgoing SIP Reguest URI and To headers with the escape sign 23 Note This parameter is applicable only if the parameter IsSpecialDigits is set 1 Defines the representation for special digits and that are used for out of band DTMF signaling using SIP INFO NOTIFY 0 Special Uses the strings and ff default 1 Numeric Uses the numerical values 10 and 11 6 7 6 Digit Collection and Dial Plan Parameters The digit collection and dial plan parameters are described in the table below Table 6 32 Digit Collection and Dial Plan Parameters Parameter Web EMS Dial Plan Index DialPlanlndex Web Digit Mapping Rules EMS Digit Map Patterns DigitMapping Version 6 0 Description Determines the Dial Plan index to use in the external Dial Plan file The Dial Plan file is loaded to the device as a dat file converted using the DConvert utility The Dial Plan index can be defined globally or per Tel Profile The valid value range is 0 to 7 where 0 denotes PLANT 1 denotes PLAN2 and so on The default is 1 indicating that no Dial Plan file is used Notes If this parameter is configured to select a Dial Plan index the settings of the parameter DigitMapping are ignored For a detailed description of the Dial Plan file refer to External Dial Plan File on page 380 Defines the digit map pattern If the digit string i e dialed number matches one of the p
324. e selects the time period by comparing the device s current time to the end time of each time period of the selected Charge Code The device generates the Number of Pulses on Answer once the call is connected and from that point on it generates a pulse each Pulse Interval If a call starts at a certain time period and crosses to the next the information of the next time period is used 3 Click the Submit button to save your changes 4 To save the changes to the flash memory refer to Saving Configuration on page 161 Version 6 0 113 March 2010 7a L tall AudioCodes MediaPack Series 3 3 4 6 5 Configuring Keypad Features The Keypad Features page enables you to activate and deactivate the following features directly from the connected telephone s keypad m Call Forward refer to Configuring Call Forward on page 140 Caller ID Restriction refer to Configuring Caller Display Information on page 138 Hotline refer to Configuring Automatic Dialing on page 137 Call Transfer Call Waiting refer to Configuring Call Waiting on page 142 Rejection of Anonymous Calls The Keypad Features page is available only for FXS interfaces The method used by the device to collect dialed numbers is identical to the method used during a regular call i e max digits interdigit timeout digit map etc The activation of each feature remains in effect until it is deactivated i e not deactivated after a call gt
325. e specified phone number Version 6 0 297 March 2010 ca AudioCodes Parameter MediaPack Series Description For example Below configuration forwards calls originally destined to Port 1 to 1001 upon On Busy Fwdinfo 0 1 1001 30 Below configuration forwards calls originally destined to Port 2 to an IP address upon On Busy Fwdinfo 1 1 2003 10 5 1 1 30 Notes Ensure that the Call Forward feature is enabled default for the settings of this table parameter to take effect To enable Call Forwarding use the parameter EnableForward Ifthe parameter Fwdlnfo Destination only contains a telephone number and a Proxy isn t used the forward to phone number must be specified in the Tel to IP Routing Prefix ini file parameter For configuring this table using the Web interface refer to Configuring Call Forward on page 140 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Call Forward Reminder Ring Parameters Notes These parameters are applicable only to FXS interfaces Fora description of this feature refer to Call Forward Reminder Ring on page 414 Web Enable NRT Subscription EnableNRTSubscription Web AS Subscribe IPGroupID ASSubscribelPGroupID Web NRT Retry Subscription Time NRTRetrySubscriptionTime Web Call Forward Ring Tone ID CallForwardRingTonelD SIP User s Manual Enables Endpoi
326. eb Based Management IP Group the call is routed to the Proxy Set IP address associated with the IP Group If the number dialed does not match these characteristics the call is not made When using a proxy server you don t need to configure this table unless you require one of the following m Fallback routing if communication is lost with proxy servers m P Security feature enabled using the SecureCallFromIP parameter the device accepts only received calls whose source IP address is defined in this routing table m Filter Calls to IP feature the device checks this routing table before a call is routed to the proxy However if the number is not allowed i e the number does not exist in the table or a Call Restriction see below routing rule is applied the call is released m Obtain different SIP URI host names per called number m Assign IP Profiles Note that for this table to take precedence over a proxy for routing calls you need to set the parameter PreferRouteTable to 1 The device checks the Destination IP Address field in this table for a match with the outgoing call A proxy is used only if a match is not found Possible uses for configuring routing rules in this table in addition to those listed above when using a proxy include the following Version 6 0 Call Restriction rejects all outgoing calls whose routing rule is associated with the destination IP address 0 0 0 0 Always Use Routing Table feature
327. ectPhonNum Redirection Phone Number 9 14 RTP Multiplexing ThroughPacket The device supports a proprietary method to aggregate RTP streams from several channels This reduces the bandwidth overhead caused by the attached Ethernet IP UDP and RTP headers and reduces the packet data transmission rate This option reduces the load on network routers and can typically save 50 e g for G 723 on IP bandwidth RTP Multiplexing ThroughPacket is accomplished by aggregating payloads from several channels that are sent to the same destination IP address into a single IP packet RTP multiplexing can be applied to the entire device refer to Configuring the RTP RTCP Settings on page 63 or to specific IP destinations using the IP Profile feature refer to Configuring IP Profiles on page 107 To enable RTP Multiplexing set the parameter RemoteBaseUDPPort to a non zero value Note that the value of RemoteBaseUDPPort on the local device must equal the value of BaseUDPPort of the remote device The device uses these parameters to identify and distribute the payloads from the received multiplexed IP packet to the relevant channels In RTP Multiplexing mode the device uses a single UDP port for all incoming multiplexed packets and a different port for outgoing packets These ports are configured using the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort When RTP Multiplexing is used call statistics are unavailable since there is no RTC
328. ed this parameter invokes the keep alive trap and sends it every 9 10 of the time defined in the parameter defining NAT Binding Default Timeout 0 Disable 1 Enable Note For this parameter to take effect a device reset is required SNMPSysOid Defines the base product system OID The default is eSNMP_AC_PRODUCT_BASE_OID_D Note For this parameter to take effect a device reset is required SNMPTrapEnterpriseOid Defines a Trap Enterprise OID The default is eSNMP_AC_ENTERPRISE_OID The inner shift of the trap in the AcTrap subtree is added to the end of the OID in this parameter Note For this parameter to take effect a device reset is required SIP User s Manual 242 Document LTRT 65413 SIP User s Manual Parameter acUserlnputAlarmDescription acUserlnputAlarmSeverity AlarmHistoryTableMaxSize Web SNMP Trap Destination Parameters EMS Network gt SNMP Managers Table 6 Configuration Parameters Reference Description Defines the description of the input alarm Defines the severity of the input alarm Determines the maximum number of rows in the Alarm History table This parameter can be controlled by the Config Global Entry Limit MIB located in the Notification Log MIB The valid range is 50 to 100 The default value is 100 Note For this parameter to take effect a device reset is required Note Up to five SNMP trap managers can be defined SNMP Manager SNMPManagerlsU
329. ed stream If a new packet arrives whose source IP address or UDP port are different to the currently accepted RTP stream one of the following occurs m The device reverts to the new RTP stream when the new packet has a source IP address and UDP port that are the same as the remote IP address and UDP port that were stated during the opening of the channel m The packet is dropped when the new packet has any other source IP address and UDP port SIP User s Manual 446 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities 10 5 10 6 Multiple Routers Support Multiple routers support is designed to assist the device when it operates in a multiple routers network The device learns the network topology by responding to Internet Control Message Protocol ICMP redirections and caches them as routing rules with expiration time When a set of routers operating within the same subnet serve as devices to that network and intercommunicate using a dynamic routing protocol the routers can determine the shortest path to a certain destination and signal the remote host the existence of the better route Using multiple router support the device can utilize these router messages to change its next hop and establish the best path Note Multiple Routers support is an integral feature that doesn t require configuration Simple Network Time Protocol Support The Simple Network Time Protocol SNTP client functionality gener
330. ed v FXS Lee P alowed FRS Ronaldino E Restricted v Allowed Allowed Allowed Allowed In the Caller ID Name field corresponding to the desired port enter the Caller ID string up to 18 characters From the Presentation drop down list select one of the following e Allowed 0 sends the string defined in the Caller ID Name field when a Tel to IP call is made using the corresponding device port e Restricted 1 the string defined in the Caller ID Name field is not sent Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 When FXS ports receive Private or Anonymous strings in the From header they don t send the calling name or number to the Caller ID display If Caller ID name is detected on an FXO line EnableCallerlD 1 it is used instead of the Caller ID name defined on this page When the Presentation field is set to Restricted the Caller ID is sent to the remote side using only the P Asserted Identity and P Preferred Identity headers AssertedldMode To maintain backward compatibility when the strings Private or Anonymous are entered in the Caller ID Name field the Caller ID is restricted and the value in the Presentation field is ignored The value of the Presentation field can be overridden by configuring the Presentation
331. edia Premium QoS EMS Premium Service Class Media Diff Serv PremiumServiceClassMediaDiffServ Version 6 0 Defines the Differentiated Services DiffServ value for Network CoS content The valid range is 0 to 63 The default value is 48 Defines the DiffServ value for Premium Media CoS content only if IPDiffServ is not set in the selected IP Profile The valid range is 0 to 63 The default value is 46 Note The value for the Premium Control DiffServ is determined by the following according to priority PDiffServ value in the selected IP Profile a PremiumServiceClassMediaDiffServ 213 March 2010 ca AudioCodes Parameter Web Control Premium QoS EMS Premium Service Class Control Diff Serv PremiumServiceClassControlDiffServ Web Gold QoS EMS Gold Service Class Diff Serv GoldServiceClassDiffServ Web Bronze QoS EMS Bronze Service Class Diff Serv BronzeServiceClassDiffServ 6 1 5 NAT and STUN Parameters MediaPack Series Description Defines the DiffServ value for Premium Control CoS content only if ControllPDiffserv is not set in the selected IP Profile The valid range is 0 to 63 The default value is 40 Notes The value for the Premium Control DiffServ is determined by the following according to priority v ControlPDiffserv value in the selected IP Profile v PremiumServiceClassControlDiffServ The same value must be configured for this parameter and the parameter MLPPDifSer
332. efer to Configuring Redirect Number Tel to IP on page 120 Phone Context refer to Mapping NPI TON to SIP Phone Context on page 122 3 3 4 7 1 Configuring General Settings The General Settings page allows you to configure general manipulation parameters For a description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt To configure the general manipulation parameters 1 Yv 2 3 4 Version 6 0 Open the General Settings page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt General Settings page item Figure 3 73 General Settings Page Set TEL to IP Redirect Reason Not Configured Configure the parameters as required Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 115 March 2010 A tal AudioCodes MediaPack Series 3 3 4 7 2 Configuring the Number Manipulation Tables The device provides four number manipulation tables for incoming IP to Tel and outgoing Tel to IP calls These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly For example telephone number manipulation can be implemented for the following E Stripping or adding dialing plan digits from or to the number respectively For example a user may need to first dial 9 before dialing the phone
333. efer to Registration Status on page 181 gt To configure the Proxy amp Registration parameters 1 Open the Proxy amp Registration page Configuration tab gt Protocol Configuration menu gt Proxies Registration IP Groups submenu gt Proxy amp Registration page item Figure 3 62 Proxy amp Registration Page Use Default Proxy Proxy Name Redundancy Mode Proxy IP List Refresh Time Enable Fallback to Routing Table Prefer Routing Table Always Use Proxy Redundant Routing Mode SIP ReRouting Mode Enable Registration Gateway Name Gateway Registration Name DNS Query Type Proxy DNS Query Type Subscription Mode Number of RTX Before Hot Swap Use Gateway Name for OPTIONS User Name Password Cnonce Authentication Mode Set Out Of Service On Registration Failure Challenge Caching Mode Mutual Authentication Mode No Parking 60 Disable No Disable Routing Table Standard Mode Disable 4 Record AR ecord Per Endpoint 3 No Default_Passwd_ Default_Cnonce Per Endpoint Enable None Optional Configure the parameters as required 3 Click the Submit button to save your changes or click the Register or Un Register buttons to save your changes and register unregister to a Proxy Registrar 4 To save the
334. eference Manual Defines the name of a private key file in unencrypted PEM format to be loaded from the TFTP server Defines the name of the HTTPS server certificate file to be loaded using TFTP The file must be in base64 encoded PEM format The valid range is a 47 character string Note This parameter is only applicable when the device is loaded using BootP TFTP For information on loading this file using the Web interface refer to the Product Reference Manual The Secure Real Time Transport Protocol SRTP parameters are described in the table below Parameter Web Media Security EMS Enable Media Security EnableMediaSecurity SIP User s Manual Table 6 20 SRTP Parameters Description Enables Secure Real Time Transport Protocol SRTP 0 Disable SRTP is disabled default 1 Enable SRTP is enabled Notes For this parameter to take effect a device reset is required SRTP reduces the number of available channels MP 124 18 available channels MP 118 6 available channels MP 114 3 available channels MP 112 No reduction AASA 234 Document LTRT 65413 SIP User s Manual Parameter Web EMS Media Security Behavior MediaSecurityBehaviour Web Master Key Identifier MKI Size EMS Packet MKI Size SRTPTxPacketMKISize Web EMS SRTP offered Suites SRTPofferedSuites Web Disable Authentication On Transmitted RTP Packets EMS RTP AuthenticationDisable Tx RTPAuthentica
335. efined Asterisks and number signs can be specified as part of the prefix Numeric ranges are allowed in the prefix m A numeric range is allowed in the number of additional digits The prefixes must not overlap Attempting to process an overlapping configuration by the DConvert utility results in an error message specifying the problematic line For a detailed description on working with Dial Plan files refer to External Dial Plan File on page 380 Version 6 0 373 March 2010 AM gA AudioCodes MediaPack Series An example of a Dial Plan file in ini file format i e before converted to dat that contains two dial plans is shown below Example of dial plan configuration This file contains two dial plans PLAN1 Defines cellular VoIP area codes 052 054 and 050 In these area codes phone numbers have 8 digits 05278 054 8 050 8 Defines International prefixes 00 012 014 The number following these prefixes may be 7 to 14 digits in length 00 7 14 O12 7 14 014 7 14 Defines emergency number 911 No additional digits are expected S PLAN2 Defines area codes 02 03 04 In these area codes phone numbers have 7 digits 0 2 4 7 Operator services starting with a star 41 42 43 No additional digits are expected 4 1 3 0 8 4 User Information File The User Information file is a text file that maps PBX extensions connected to t
336. el routing table configure the parameter RedundantRoutingMode to 1 default The reasons for alternative routing for Tel to IP calls also apply for Proxies if the parameter RedundantRoutingMode is set to 2 You can also configure alternative routing using the ini file table parameters AltRouteCauseTel2IP and AltRouteCauselP2Tel refer to Number Manipulation and Routing Parameters on page 331 Version 6 0 125 March 2010 7a K tal AudioCodes MediaPack Series 3 3 4 8 2 Configuring General Routing Parameters The Routing General Parameters page allows you to configure the general routing parameters For a description of these parameters refer to Configuration Parameters Reference on page 207 gt To configure the general routing parameters 1 Open the Routing General Parameters page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Routing General Parameters page item Figure 3 78 Routing General Parameters Page Add Hunt Group ID as Prefix Add Trunk ID as Prefix Replace Empty Destination with B channel Phone Number Add NPI and TON to Called Number Add NPI and TON to Calling Number IP to Tel Remove Routing Table Prefix Source IP Address Input SIP Contact Header Enable Alt Routing Tel to IP Disable Alt Routing Tel to IP Mode Both Alt Routing Tel to IP Connectivity Method ICMP Ping SSSR SNR NR Alt Routing Te
337. el to IP Routing on page 126 Note If the parameter PreferRouteTable is set to 1 refer to Configuring Proxy and Registration Parameters on page 96 the routing rules in the Outbound IP Routing Table prevail over the selected Serving IP Group ID The host name used in the SIP From header in INVITE messages and as a host name in From To headers in REGISTER requests If not configured the global parameter SIPGatewayName is used instead The user part in the SIP Contact URI in INVITE messages and as a user part in From To and Contact headers in REGISTER requests This is applicable only if the field Registration Mode is set to Per Account and the Registration through the Account table is successful Notes f registration fails then the userpart in the INVITE Contact header contains the source party number The ContactUser parameter in the Account Table page overrides this parameter 87 March 2010 7a c tall AudioCodes MediaPack Series 3 3 43 Protocol Definition The Protocol Definition submenu allows you to configure the main SIP protocol parameters This submenu contains the following page items m SIP General Parameters refer to SIP General Parameters on page 88 m DTMF 8 Dialing refer to DTMF amp Dialing Parameters on page 90 3 3 4 3 1 Configuring SIP General Parameters The SIP General Parameters page is used to configure general SIP parameters For a description of the
338. eld enter the fully qualified DNS name FQDN as the certificate subject and then click Generate Self signed after a few seconds a message appears displaying the new subject name Save configuration refer to Saving Configuration on page 161 and then restart the device for the new certificate to take effect SIP User s Manual 76 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 3 5 Configuring the 802 1x Settings The 802 1x Settings page is used to configure IEEE 802 1X LAN security The device can function as an IEEE 802 1X supplicant IEEE 802 1X is a standard for port level security on secure Ethernet switches when a device is connected to a secure port no traffic is allowed until the identity of the device is authenticated The device supports the following Extensible Authentication Protocol EAP variants m MD5 Challenge EAP MD5 m Protected EAP PEAPVO with EAP MSCHAPv2 m EAP TLS For a description of the parameters appearing on this page refer Configuration Parameters Reference on page 207 For a detailed description of this feature refer to the Product Reference Manual gt To configure the 802 1x parameters 1 Open the 802 1x Settings page Configuration tab gt Security Settings menu gt 802 1x Settings page item Figure 3 52 8021x Settings Page 802 1x Mode Disabled 802 1x Username 802 1x Password o 802 1x Verify Peer Certificate Disable
339. emium Media Premium Control Gold and Bronze The DiffServ parameters are described in Networking Parameters on page 207 Network Configuration The device allows you to configure up to 16 different IP addresses with associated VLANs using the Multiple Interface table In addition complementing this table is the Routing table which allows you to define routing rules for non local hosts subnets This section describes the various network configuration options offered by the device Multiple Network Interfaces and VLANs A need often arises to have logically separated network segments for various applications for administrative and security reasons This can be achieved by employing Layer 2 VLANs and Layer 3 subnets Figure 10 2 Multiple Network Interfaces AudioCodes Media Gateway Network Internet Separated Networks Scheme This figure above depicts a typical configuration featuring in which the device is configured with three network interfaces for m Operations Administration Maintenance and Provisioning OAMP applications m Call Control applications m Media SIP User s Manual 448 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities It is connected to a VLAN aware switch which is used for directing traffic from and to the device to three separated Layer 3 broadcast domains according to VLAN tags middle pane The Multiple Interfaces scheme allows the configuration of up to 16 dif
340. en Suffix2Add The following parameters are not applicable NumberType NumberPlan and IsPresentationRestricted Determines whether the received Phone Context parameter is added as a prefix to the outgoing Called and Calling numbers 0 Disable Disable default 1 Enable Enable 342 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web Phone Context Table Description EMS SIP Manipulations gt Phone Context PhoneContext Web EMS Add Hunt Group ID as Prefix AddTrunkGroupAsPrefix Version 6 0 This ini file table parameter defines the Phone Context table This parameter maps NPI and TON to the SIP Phone Context parameter When a call is received from the Tel the NPI and TON are compared against the table and the corresponding Phone Context value is used in the outgoing SIP INVITE message The same mapping occurs when an INVITE with a Phone Context attribute is received The Phone Context parameter appears in the standard SIP headers Request URI To From Diversion where a phone number is used The format for this parameter is as follows PhoneContext FORMAT PhoneContext_Index PhoneContext_Npi PhoneContext_Ton PhoneContext_Context PhoneContext For example PhoneContext 0 0 0 unknown com PhoneContext 1 1 1 host com PhoneContext 2 9 1 na e164 host com Notes This parameter can include up to 20 indices Several entries wi
341. ence Level in Resource Priority SIP Header 0 lowest routine 2 priority 4 immediate 6 flash 8 flash override 9 highest flash override override Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Routine precedence call level Routine The valid range is 1 to 63 The default is 1 MEPPROunNeRTPDStr Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call using the parameter IPProfile Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Priority precedence call level Priority The valid range is 1 to 63 The default is 1 MEP PEnOnYRIERSCE Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call using the parameter IPProfile Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Immediate precedence call Immediate level The valid range is 1 to 63 The default is 1 IMEPPimmediateRtPOSCF Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call using the parameter IPProfile Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash precedence call level Flash The valid range is 1 to 63 The default is 1 MERE Flas ittepscr Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffS
342. encyPrefix MediaPack Series Description 2 Ignore Register Use regular SAS Normal Emergency logic same as option 0 but when in Normal mode incoming REGISTER requests are ignored 3 Auto answer REGISTER When in Normal mode the device responds to received REGISTER requests by sending a SIP 200 OK instead of relaying the registration requests to a Proxy and enters the registrations in its SAS database Determines the SAS application database binding mode 0 URI If the incoming AoR in the INVITE requests is using a tel URI or user phone is defined the binding is performed according to the user part of the URI only Otherwise the binding is according to the entire URI i e User Host default 1 User Part only The binding is always performed according to the User Part only Defines emergency numbers for the device s SAS application When the device s SAS agent receives a SIP INVITE from an IP phone that includes one of the emergency numbers in the SIP user part it forwards the INVITE to the default gateway configured by the parameter SASDefaultGatewaylP i e the device itself which sends the call directly to the PSTN This is important for routing emergency numbers such as 911 in North America directly to the PSTN This is applicable to SAS operating in Normal and Emergency modes Up to four emergency numbers can be defined where each number can be up to four digits Define
343. end with a semicolon m End of Table Mark Indicates the end of the table The same string used for the table s title preceded by a backslash V e g MY TABLE NAME SIP User s Manual 186 Document LTRT 65413 SIP User s Manual 4 INI File Configuration The following displays an example of the structure of an ini file table parameter Table Title lnc SEANEME OOP Necale FORMAT Index Column Namel Column Name2 Column Name3 This is the Format line Index 0 valuel value2 value3 Index 1 valuel value3 These are the Data lines NTable Title This is the end of the table mark The ini file table parameter formatting rules are listed below m Indices in both the Format and the Data lines must appear in the same order The Index field must never be omitted m The Format line can include a subset of the configurable fields in a table In this case all other fields are assigned with the pre defined default values for each configured line m The order of the fields in the Format line isn t significant as opposed to the Index fields The fields in the Data lines are interpreted according to the order specified in the Format line m The double dollar sign in a Data line indicates the default value for the parameter The order of the Data lines is insignificant Data lines must match the Format line i e it must contain exactly the same number of Indices and Data fields and must be i
344. ends and the tone detection is reported For continuous tones this parameter is ignored e Second Signal On Time 10 msec Signal On period in 10 msec units for the second cadence on off cycle Can be omitted if there isn t a second cadence e Second Signal Off Time 10 msec Signal Off period in 10 msec units for the second cadence on off cycle Can be omitted if there isn t a second cadence e Third Signal On Time 10 msec Signal On period in 10 msec units for the third cadence on off cycle Can be omitted if there isn t a third cadence e Third Signal Off Time 10 msec Signal Off period in 10 msec units for the third cadence on off cycle Can be omitted if there isn t a third cadence e Fourth Signal On Time 10 msec Signal On period in 10 msec units for the fourth cadence on off cycle Can be omitted if there isn t a fourth cadence e Fourth Signal Off Time 10 msec Signal Off period in 10 msec units for the fourth cadence on off cycle Can be omitted if there isn t a fourth cadence e Carrier Freq Hz Frequency of the carrier signal for AM tones e Modulation Freq Hz Frequency of the modulated signal for AM tones valid range from 1 to 128 Hz e Signal Level dBm Level of the tone for AM tones e AM Factor steps of 0 02 Amplitude modulation factor valid range from 1 to 50 Recommended values from 10 to 25 When the same frequency is used for a continuous tone and a cadence tone
345. ents a line of text in the Welcome message box Up to 20 indices can be defined The configured text message must be enclosed in double quotation marks i e If this parameter is not configured no Welcome message is displayed For a description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Telnet Parameters The Telnet parameters are described in the table below Parameter Web Embedded Telnet Server EMS Server Enable TelnetServerEnable Web Telnet Server TCP Port EMS Server Port TelnetServerPort Web Telnet Server Idle Timeout EMS Server Idle Disconnect TelnetServerldleDisconnect SIP User s Manual Table 6 12 Telnet Parameters Description Enables or disables the device s embedded Telnet server Telnet is disabled by default for security 0 Disable default 1 Enable Unsecured 2 Enable Secured SSL Note Only the primary Web User Account which has Security Administration access level can access the device using Telnet refer to Configuring the Web User Accounts on page 66 Defines the port number for the embedded Telnet server The valid range is all valid port numbers The default port is 23 Defines the timeout in minutes for disconnection of an idle Telnet session When set to zero idle sessions are not disconnected The valid range is any value The default value is 0 Note For this parameter to take effect a dev
346. er FXO FXS device The device forwards calls using this table only if no alternative IP to Tel routing has been configured or alternative routing fails and the following call forward reason included in the SIP Diversion header of 3xx messages exists unavailable All FXO FXS lines pertaining to a Hunt Group are busy or unavailable The format of this parameter is as follows ForwardOnBusyTrunkDest FORMAT ForwardOnBusyTrunkDest Index ForwardOnBusyTrunkDest TrunkGroupld ForwardOnBusyTrunkDest ForwardDestination ForwardOnBusyTrunkDest SIP User s Manual 340 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description For example the below configuration forwards IP to Tel calls to destination IP address 10 13 4 12 port 5060 using transport protocol TCP if Hunt Group ID 2 is busy ForwardOnBusyTrunkDest 1 2 10 13 4 12 5060 transport tcp Notes The maximum number of indices starting from 1 depends on the maximum number of Hunt Groups For the destination instead of a dotted decimal IP address FQDN can be used In addition the following syntax can be used host port transport xxx i e IP address port and transport type 6 15 3 Number Manipulation Parameters The number manipulation parameters are described in the table below Table 6 56 Number Manipulation Parameters Parameter Web Copy Destination Number to Redirect Number EMS Copy Dest to
347. er of attempted calls It is composed of established and failed calls The number of established calls is represented by the Number of Established Calls counter The number of failed calls is represented by the failed call counters Only one of the established failed call counters is incremented every time Indicates the number of established calls It is incremented as a result of one of the following release reasons if the duration of the call is greater than zero GWAPP_REASON_NOT_RELEVANT 0 GWAPP_NORMAL_CALL_CLEAR 16 GWAPP_NORMAL_UNSPECIFIED 31 And the internal reasons RELEASE BECAUSE UNKNOWN REASON RELEASE BECAUSE REMOTE CANCEL CALL RELEASE BECAUSE MANUAL DISC RELEASE BECAUSE SILENCE DISC RELEASE BECAUSE DISCONNECT CODE Note When the duration of the call is zero the release reason GWAPP NORMAL CALL CLEAR increments the Number of Failed Calls due to No Answer counter The rest of the release reasons increment the Number of Failed Calls due to Other Failures counter The percentage of established calls from attempted calls Indicates the number of calls that failed as a result of a busy line It is incremented as a result of the following release reason GWAPP_USER_BUSY 17 Indicates the number of calls that weren t answered It s incremented as a result of one of the following release reasons GWAPP_NO_USER_RESPONDING 18 GWAPP NO ANSWER FROM USER ALERTED 19 GWAPP NORMAL CALL CLEAR
348. erTime SIP User s Manual Table 6 59 DTMF Parameters Description Determines the DTMF transport type 0 DTMF Mute Erases digits from voice stream and doesn t relay to remote 2 Transparent DTMF Digits remain in voice stream 3 RFC 2833 Relay DTMF Erases digits from voice stream and relays to remote according to RFC 2833 default 7 RFC 2833 Relay Rev Mute DTMFs are sent according to RFC 2833 and muted when received Note This parameter is automatically updated if the parameters TxDTMFOption or RXxDTMFOption are configured DTMF gain control value in decibels to the or analog side The valid range is 31 to 0 dB The default value is 11 dB Defines the range in decibels between the high and low frequency components in the DTMF signal Positive decibel values cause the higher frequency component to be stronger than the lower one Negative values cause the opposite effect For any parameter value both components change so that their average is constant The valid range is 10 to 10 dB The default value is 0 dB Note For this parameter to take effect a device reset is required Time in msec between generated DTMF digits to PSTN side if TxDTMFOption 1 2 or 3 The default value is 100 msec The valid range is 0 to 32767 Time in msec for generating DTMF tones to the PSTN side if TxDTMFOption 1 2 or 3 It also configures the duration that is sent in INFO Cisco messages The val
349. ere 0 denotes Port 1 IsEnable Enables 1 or disables 0 default Japan NTT Modem DID support For example EnableDID 0 1 DID is enabled on Port 1 Notes This parameter is applicable only to FXS interfaces Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 307 March 2010 ca AudioCodes MediaPack Series Parameter Description WinkTime Defines the time in msec elapsed between two consecutive polarity reversals This parameter can be used for DID signaling The valid range is 0 to 4 294 967 295 The default is 200 Notes This parameter is applicable to FXS and FXO interfaces For this parameter to take effect a device reset is required 6 8 12 MLPP Parameters The Multilevel Precedence and Preemption MLPP parameters are described in the table below Parameter Web EMS Call Priority Mode CallPriorityMode Web MLPP DiffServ EMS Diff Serv MLPPDiffserv EMS E911 MLPP Behavior E911MLPPBehavior Web EMS Precedence Ringing Type PrecedenceRingingType SIP User s Manual Table 6 45 MLPP Parameters Description Enables MLPP Priority Call handling 0 Disable Disable default 1 MLPP Priority Calls handling is enabled Defines the DiffServ value differentiated services code point DSCP used in IP packets containing SIP messages that are related to MLPP calls This parameter defines DiffSer
350. erform the following procedure a Enable DHCP and save the configuration b Perform a cold reset using the device s hardware reset button soft reset using the Web interface doesn t trigger the BootP DHCP procedure and this parameter reverts to Disable Throughout the DHCP procedure the BootP TFTP application must be deactivated otherwise the device receives a response from the BootP server instead of from the DHCP server For additional information on DHCP refer to the Product Reference Manual This parameter is a special Hidden parameter Once defined and saved in flash memory its assigned value doesn t revert to its default even if the parameter doesn t appear in the ini file EMS DHCP Speed Factor Determines the DHCP renewal speed DHCPSpeedFactor 0 Disable 1 Normal default 2 to 10 Fast When set to 0 the DHCP lease renewal is disabled Otherwise the renewal time is divided by this factor Some DHCP enabled routers perform better when set to 4 Note For this parameter to take effect a device reset is required Web Enable DHCP Lease Enables or disables DHCP renewal support Renewal EnableDHCPLeaseRenewal Die san 1 Enable This parameter is applicable only if the parameter DHCPEnable is set to 0 for cases where booting up the device using DHCP is not desirable but renewing DHCP leasing is When the device is powered up it attempts to communicate with a BootP server If
351. ers Group 0 tab the Coders screen is displayed a b c Click the lagi button to add a new Coder entry and then click Yes to confirm Double click each field to enter values Right click the new entry and then choose Unlock Rows 4 Select the Proxy Server tab a b SIP User s Manual Set Proxy Used to Yes Optional In the Proxy Name field enter the Proxy s name The Proxy name replaces the Proxy IP address in all SIP messages This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead When no Proxy is used the internal routing table is used to route the calls Click the button and then click Yes to confirm Enter the IP address of the Proxy Server Right click the new entry and then choose Unlock Rows 196 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS Version 6 0 Select the Registration tab a Configure Is Register Needed field No the device doesn t register to a Proxy Registrar server default Yes the device registers to a Proxy Registrar server at power up and every user defined interval Registration Time parameter b Click Apply and close the active window Open the SIP EndPoints frame Configuration icon gt SIP Endpoints menu a Click the button to add a new entry and then click Yes to confirm the Phones screen is displayed b Double click each field to enter values
352. ers Reference 6 17 Auxiliary and Configuration Files Parameters This subsection describes the device s auxiliary and configuration files parameters 6 17 1 Auxiliary Configuration File Name Parameters The configuration files i e auxiliary files can be loaded to the device using the Web interface or a TFTP session refer to Loading Auxiliary Files on page 163 For loading these files using the ini file you need to configure these files in the ini file and configured whether they must be stored in the non volatile memory The table below lists the ini file parameters associated with these auxiliary files For a detailed description of the auxiliary files refer to Auxiliary Configuration Files on page 367 Table 6 61 Auxiliary and Configuration File Parameters Parameter Description General Parameters SetDefaultOnIniFileProcess Determines if all the device s parameters are set to their defaults before processing the updated ini file 0 Disable parameters not included in the downloaded ini file are not returned to default settings i e retain their current settings 1 Enable default Note This parameter is applicable only for automatic HTTP update or Web ini file upload not applicable if the ini file is loaded using BootP SaveConfiguration Determines if the device s configuration parameters and files is saved to flash non volatile memory 0 Configuration isn t saved to flash memory
353. erv or as defined for IP Profiles per call using the parameter IPProfile Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash Override precedence Flash Override call level MLPPFlashOverRTPDSCP The valid range is 1 to 63 The default is 1 Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call using the parameter IPProfile Web EMS RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash Override Override Flash Override Override precedence call level MLPPFlashOverOverRTPDSCP The valid range is 1 to 63 The default is 1 Note If set to 1 the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call using the parameter IPProfile Version 6 0 309 March 2010 ca AudioCodes 6 9 MediaPack Series Standalone Survivability Parameters The Stand alone Survivability SAS parameters are described in the table below Parameter Web Enable SAS EMS Enable EnableSAS Web SAS Local SIP UDP Port EMS Local SIP UDP SASLocalSIPUDPPort Web SAS Default Gateway IP EMS Default Gateway IP SASDefaultGatewaylP Web SAS Registration Time EMS Registration Time SASRegistrationTime Web SAS Local SIP TCP Port EMS Local SIP TCP Port SASLocalSIPTCPPort Web SAS Local SIP TLS Port EMS Local SIP TLS Port SASLocalSIPTLSPort Web EMS Enable
354. es To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 17 Description of Parameter Unique to IP Profile Parameter Description Number of Calls Limit Maximum number of concurrent calls If the profile is set to some limit the 3 3 4 6 3 3 4 6 1 device maintains the number of concurrent calls incoming and outgoing pertaining to the specific profile A limit value of 1 indicates that there is no limitation on calls for that specific profile default A limit value of 0 indicates that all calls are rejected When the number of concurrent calls is equal to the limit the device rejects any new incoming and outgoing calls belonging to that profile SIP Advanced Parameters The SIP Advanced Parameters submenu allows you to configure advanced SIP control protocol parameters This submenu contains the following page items Advanced Parameters refer to Configuring Advanced Parameters on page 109 Supplementary Services refer to Configuring Supplementary Services on page 111 Metering Tones refer to Configuring Metering Tones on page 112 Charge Codes refer to Configuring the Charge Codes Table on page 113 Keypad Features refer to Configuring Keypad Features on page 114 Configuring Advanced Parameters The Advanced Parameters page allows you to configure advanced SIP control parameters For a description of the parameters appearing on this page refer to Confi
355. es corresponding to the parameters c Click Next 5 After clicking Next a message box appears notifying you of the change Click OK 6 Click Save amp Finish a message box appears informing you that the Scenario has been successfully modified The Scenario mode is exited and the menus of the Configuration tab appear in the Navigation tree 3 1 8 4 Saving a Scenario to a PC You can save a Scenario to a PC as a dat file This is especially useful when requiring more than one Scenario to represent different environment setups e g where one includes PBX interoperability and another not Once you create a Scenario and save it to your PC you can then keep on saving modifications to it under different Scenario file names When you require a specific network environment setup you can simply load the suitable Scenario file from your PC refer to Loading a Scenario to the Device on page 42 gt To save a Scenario to a PC 1 On the Navigation bar click the Scenarios tab the Scenario appears in the Navigation tree 2 Click the Get Send Scenario File button located at the bottom of the Navigation tree the Scenario File page appears as shown below Figure 3 18 Scenario File Page Seomario Fle Get the Scenario file from the device to your computer Get Scenario File Send Scenario file from your computer to the device Browse Send File Version 6 0 41 March 2010 7a c tall AudioCodes MediaPack Series
356. es the appropriate extension when messages and the number of messages are pending The FXO device detects an MWI message from the Tel PBX side using any one of the following methods m 100 VDC sent by the PBX to activate the phone s lamp Stutter dial tone from the PBX m MWI display signal according to the parameter CallerlDType Version 6 0 395 March 2010 Aa c AudioCodes MediaPack Series Upon detection of an MWI message the FXO device sends a SIP NOTIFY message to the IP side When receiving this NOTIFY message the remote FXS device generates an MWI signal toward its Tel side Figure 9 9 MWI for Remote Extensions FXO Device gt FXS Device Remote PBX Extensions 9 4 3 4 Call Waiting for Remote Extensions When the FXO device detects a Call Waiting indication FSK data of the Caller Id CalleriIDType2 from the PBX it sends a proprietary INFO message which includes the caller identification to the FXS device Once the FXS device receives this INFO message it plays a call waiting tone and sends the caller ID to the relevant port for display The remote extension connected to the FXS device can toggle between calls using the Hook Flash button Figure 9 10 Call Waiting for Remote Extensions FXO Device FXS Device Remote PBX Extensions Plays Call Waiting Tone and Sends Caller ID SIP User s Manual 396 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 4 3 5 FXS
357. essages when the parameter EnableProxyKeepAlive is set to 2 REGISTER Typically the device registers every 3 600 sec i e one hour The device resumes registration according to the parameter RegistrationTimeDivider The valid range is 10 to 2 000 000 The default value is 180 Defines the re registration timing in percentage The timing is a percentage of the re register timing set by the Registrar server The valid range is 50 to 100 The default value is 50 For example If this parameter is set to 70 and the Registration Expires time is 3600 the device re sends its registration request after 3600 x 70 i e 2520 sec Note This parameter may be overridden if the parameter RegistrationTimeThreshold is greater than 0 Defines the time interval in seconds after which a registration request is re sent if registration fails with a 4xx response or if there is no response from the Proxy Registrar server The default is 30 seconds The range is 10 to 3600 Defines a threshold in seconds for re registration timing If this parameter is greater than 0 but lower than the computed re registration timing according to the parameter RegistrationTimeDivider the re registration timing is set to the following timing set by the Registration server in the SIP Expires header minus the value of the parameter RegistrationTimeThreshold The valid range is 0 to 2 000 000 The default value is 0 Enables immediate re registration if a
358. estination IP address Once a route is located the Charge Code configured for that route is used to associate the route with an entry in the Charge Codes table To configure the Metering tones 1 Open the Metering Tones page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Metering Tones page item Figure 3 70 Metering Tones Page v Generate Metering Tones Disable Metering Tone Type 16 KHz Charge Codes Table 2 Configure the Metering tones parameters as required For a description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 3 Click the Submit button to save your changes 4 To save the changes to the flash memory refer to Saving Configuration on page 161 If you set the Generate Metering Tones parameter to Internal Table access the Charge Codes Table page by clicking the Charge Codes Table gt button For a detailed description on configuring the Charge Codes table refer to Charge Codes Table on page 113 SIP User s Manual 112 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 6 4 Configuring the Charge Codes Table The Charge Codes Table page is used to configure the metering tones and their time interval that the FXS interfaces generate to the Tel side To associate a charge code to an outgoing Tel to IP call use the Tel to IP Routing
359. estination addresses and UDP port equal the SAS feature s IP address and SAS local SIP UDP port gt To configure the Stand Alone Survivability parameters 1 Open the SAS Configuration page Configuration tab gt Protocol Configuration menu gt SAS submenu gt Stand Alone Survivability page item Figure 3 91 SAS Configuration Page SAS Local SIP UDP Port 5080 SAS Default Gateway IP SAS Registration Time 20 SAS Local SIP TCP Port SAS Local SIP TLS Port SAS Proxy Set SAS Emergency Numbers SAS Binding Mode O URI SAS Survivability Mode 1 Always Emergency Enable ENUM Disable Redundant SAS Proxy Set E SAS Registration Manipulation Remove From Right Leave From Right 0 0 v SAS Routing SAS Routing Table 2 Configure the parameters as described in SIP Configuration Parameters on page 245 3 Click the Submit button to apply your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 To configure the SAS Routing table under the SAS Routing group click the SAS Routing Table button to open the IP2IP Routing Table page For a description of this table refer to Configuring the IP2IP Routing Table SAS on page 146 Version 6 0 145 March 2010 Aa tal AudioCodes MediaPack Series 3 3 4 11 2Configuring the IP2IP Routing Table SAS The IP2IP Routing Table page allows
360. et Access List can also be defined using the ini file parameter WebAccessList x refer to Web and Telnet Parameters on page 222 gt To add authorized IP addresses for Web and Telnet interfaces access 1 Open the Web 8 Telnet Access List page Configuration tab gt Security Settings menu gt Web 8 Telnet Access List page item Figure 3 47 Web 8 Telnet Access List Page Add New Entry Add New Entry 2 To add an authorized IP address in the Add a New Authorized IP Address field enter the reguired IP address and then click Add New Address the IP address you entered is added as a new entry to the Web 8 Telnet Access List table Figure 3 48 Web amp Telnet Access List Table Delete Authorized IP Row Address Delete Selected Addresses Note Delete all rows to allow access from any IP address to WEB 8 Telnet Add New Entry Version 6 0 69 March 2010 7a e AudioCodes MediaPack Series 3 To delete authorized IP addresses select the Delete Row check boxes corresponding to the IP addresses that you want to delete and then click Delete Selected Addresses the IP addresses are removed from the table and these IP addresses can no longer access the Web and Telnet interfaces 4 To save the changes to flash memory refer to Saving Configuration on page 161 The first authorized IP address in the list must be your PC s terminal IP address otherwise access from your PC is denied
361. et to 1 Enables the use of DNS Naming Authority Pointer NAPTR and Service Record SRV queries to resolve Proxy and Registrar servers and to resolve all domain names that appear in the SIP Contact and Record Route headers 0 A Record default 1 SRV 2 NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy Registrar IP address parameter Contact Record Route headers or IP address defined in the Routing tables contain a domain name an SRV query is performed The device uses the first host name received from the SRV query The device then performs a DNS A record query for the host name to locate an IP address If set to NAPTR 2 an NAPTR query is performed If it is successful an SRV query is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy Registrar IP address parameter the domain name in the Contact Record Route headers or the IP address defined in the Routing tables contain a domain name with port definition the device performs a regular DNS A record query If a specific Transport Type is defined a NAPTR query is not performed Note To enable NAPTR SRV queries for Proxy servers only use the parameter ProxyDNSQueryType 267 March 2010 ca AudioCodes Parameter Web Proxy DNS Auery Type ProxyDNSQueryType Web EMS Use Gateway
362. etering signal pulse voltage level TTX Level EMS TTX Voltage Level AnalogTTXVoltageLevel 1 default 0 5 Vrms sinusoidal bursts 2 1 Vrms sinusoidal bursts 0 0 Vrms sinusoidal bursts Notes For this parameter to take effect a device reset is required This parameter is applicable only to FXS interfaces SIP User s Manual 322 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web Charge Codes Table EMS Charge Codes ChargeCode Version 6 0 Description This ini file table parameter configures metering tones and their time intervals that the device s FXS interface generates to the Tel side The format of this parameter is as follows ChargeCode FORMAT ChargeCode Index ChargeCode EndTime1 ChargeCode Pulselnterval1 ChargeCode_PulsesOnAnswer1 ChargeCode EndTime2 ChargeCode Pulselnterval2 ChargeCode PulsesOnAnswer2 ChargeCode EndTime3 ChargeCode Pulselnterval3 ChargeCode PulsesOnAnswer3 ChargeCode EndTime4 ChargeCode Pulselnterval4 ChargeCode PulsesOnAnswer4 ChargeCode Where EndTime Period 1 4 end time Pulselnterval Period 1 4 pulse interval PulsesOnAnswer Period 1 4 pulses on answer For example ChargeCode 1 7 30 1 14 20 2 20 15 1 0 60 1 ChargeCode 2 5 60 1 14 20 1 0 60 1 ChargeCode 3 0 60 1 ChargeCode 0 6 3 1 12 2 1 18 5 2 0 2 1 Notes The parameter can include
363. eters of this ini file table parameter refer to Configuring the Firewall Settings on page 70 Fora description of configuring with ini file table parameters refer to Configuring ini File Table Parameters on page 186 6 4 2 HTTPS Parameters The Secure Hypertext Transport Protocol HTTPS parameters are described in the table below Parameter Web Secured Web Connection HTTPS EMS HTTPS Only HTTPSOnly EMS HTTPS Port HTTPSPort EMS HTTPS Cipher String HTTPSCipherString Web HTTP Authentication Mode EMS Web Authentication Mode WebAuthMode Version 6 0 Table 6 19 HTTPS Parameters Description Determines the protocol used to access the Web interface 0 HTTP and HTTPS default 1 HTTPs Only Unencrypted HTTP packets are blocked Note For this parameter to take effect a device reset is required Determines the local Secured HTTPS port of the device The valid range is 1 to 65535 other restrictions may apply within this range The default port is 443 Note For this parameter to take effect a device reset is required Defines the Cipher string for HTTPS in OpenSSL cipher list format For the valid range values refer to URL http www openssl org docs apps ciphers html The default value is EXP Export encryption algorithms For example use ALL for all ciphers suites The only ciphers available are RC4 and DES and the cipher bit strength is limited to 56 bits
364. etization Period FaxModemBypassM SIP User s Manual MediaPack Series Description Notes The rate is negotiated between both sides i e the device adapts to the capabilities of the remote side Configuration above 14 4 kbps is truncated to 14 4 kbps for non T 38 V 34 supporting lt devices gt Determines whether the Error Correction Mode ECM mode is used during fax relay 0 Disable ECM mode is not used during fax relay 1 Enable ECM mode is used during fax relay default Coder used by the device when performing fax modem bypass Usually high bit rate coders such as G 711 should be used 0 G 711Alaw G 711 A law 64 default 1 G 711Mulaw G 711 p law Determines whether the device detects the fax Calling tone CNG 0 Disable The originating device doesn t detect CNG the CNG signal passes transparently to the remote side default 1 Relay CNG is detected on the originating side CNG packets are sent to the remote side according to T 38 if IsFaxUsed 1 and the fax session is started A SIP Re INVITE message isn t sent and the fax session starts by the terminating device This option is useful for example when the originating device is located behind a firewall that blocks incoming T 38 packets on ports that have not yet received T 38 packets from the internal network i e originating device To also send a Re INVITE message upon detection of a fax CNG tone in t
365. evice the new capabilities and resources are active If the Syslog server indicates that the Software Upgrade Key file was unsuccessfully loaded i e the SN_ line is blank perform the following preliminary troubleshooting procedures 1 Open the Software Upgrade Key file and check that the S N line appears If it does not appear contact AudioCodes 2 Verify that you ve loaded the correct file Open the file and ensure that the first line displays LicenseKeys 3 Verify that the contents of the file has not been altered in any way 3 4 2 2 1 Loading via BootP TFTP The procedure below describes how to load a Software Upgrade Key to the device using AudioCodes BootP TFTP Server utility for a detailed description on the BootP utility refer to the Product Reference Manual gt To load a Software Upgrade Key file using BootP TFTP 1 Place the Software Upgrade Key file typically a txt file in the same folder in which the device s cmp file is located 2 Start the BootP TFTP Server utility Version 6 0 167 March 2010 A tal AudioCodes MediaPack Series 3 4 2 3 From the Services menu choose Clients the Client Configuration screen is displayed From the INI File drop down list select the Software Upgrade Key file Note that the device s cmp file must be specified in the Boot File field Configure the initial BootP TFTP parameters as required and then click OK Reset the device
366. ex CallerDisplaylnfo DisplayString CallerDisplaylnfo IsCidRestricted CallerDisplayInfo Where Index Port number where 0 depicts Port 1 DisplayString Caller ID string up to 18 characters IsCidRestricted v 0 Allowed sends the defined caller ID string when a Tel to IP call is made using the corresponding device port default v 1 Restricted does not send the defined caller ID string For example CallerDisplaylnfo 0 Susan C 0 Susan C is sent as the Caller ID for Port 1 CallerDisplaylnfo 1 Mark M 0 Mark M is sent as Caller ID for Port 2 Notes When FXS ports receive Private or Anonymous strings in the SIP From header the calling name or number is not sent to the Caller ID display Ifthe Caller ID name is detected on an FXO line the parameter EnableCallerlD is set to 1 it is used instead of the Caller ID name defined in this table parameter When the parameter CallerDisplayInfo IsCidRestricted is set to 1 Restricted the Caller ID is sent to the remote side using only the SIP headers P Asserted Identity and P Preferred Identity AssertedldMode To maintain backward compatibility when the strings Private or Anonymous are entered in the parameter CallerDisplaylnfo DisplayString the Caller ID is restricted and the value of the parameter CallerDisplaylnfo IsCidRestricted is ignored The value of the parameter CallerDisplaylnfo IsCidRestric
367. ex mode of the Ethernet port Port Speed Displays the speed in Mbps of the Ethernet port 3 5 1 3 Viewing Active IP Interfaces The IP Interface Status page displays the device s active IP interfaces which are configured in the Multiple Interface Table page refer to Configuring the Multiple Interface Table on page 50 gt To view the Active IP Interfaces page m Open the IP Interface Status page Status amp Diagnostics tab gt Status amp Diagnostics menu gt IP Interface Status page item Table 3 31 IP Interface Status Page Length ame Index Application Type Address Type Interface Mode IP Address Prefix Gateway VLAN 10 rele NA O M C IPv4 Manual 10 8 7 31 SIP User s Manual 174 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 5 1 4 Viewing Device Information The Device Information page displays the device s specific hardware and software product information This information can help you expedite troubleshooting Capture the page and e mail it to AudioCodes Technical Support personnel to ensure quick diagnosis and effective corrective action This page also displays any loaded files used by the device stored in the RAM and allows you to remove them gt To access the Device Information page m Open the Device Information page Status amp Diagnostics tab gt Status 8 Diagnostics menu gt Device Information page item Table 3 32 Device Informati
368. f different interfaces must not overlap in any way e g defining two interfaces with 10 0 0 1 8 and 10 50 10 1 24 is invalid Each interface must have its own address space D3 March 2010 ca AudioCodes Parameter Web EMS Gateway InterfaceTable Gateway Web EMS VLAN ID InterfaceTable VlanlD Web EMS Interface Name InterfaceTable InterfaceName General Parameters VLAN Mode VIANMode Native VLAN ID VLANNativeVlanID MediaPack Series Description Defines the IP address of the default gateway used by the device Notes Only one default gateway can be defined The default gateway must be configured on an interface that includes Media traffic The default gateway s IP address must be in the same subnet as the interface address Apart from the interface with the defined default gateway for all other interfaces define this parameter to 0 0 0 0 For configuring additional routing rules for other interfaces use the Routing table refer to Configuring the IP Routing Table on page 58 Defines the VLAN ID for each interface Incoming traffic with this VLAN ID is routed to the corresponding interface and outgoing traffic from that interface is tagged with this VLAN ID Notes The VLAN ID must be unique for each interface VLANs are available only when booting the device from flash When booting using BootP DHCP protocols VLANs are disabled to allow easier maintenance access
369. fer to Configuring Voice Mail Parameters on page 148 m RADIUS Parameters refer to Configuring RADIUS Accounting Parameters on page 149 m FXO Settings refer to Configuring FXO Parameters on page 151 3 3 5 1 Configuring Voice Mail Parameters The Voice Mail Settings page allows you to configure the voice mail parameters For a description of these parameters refer to Configuration Parameters Reference on page 207 The Voice Mail Settings page is available only for FXO interfaces For detailed information on configuring the voice mail application refer to the CPE Configuration Guide for Voice Mail User s Manual SIP User s Manual 148 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To configure the Voice Mail parameters 1 Open the Voice Mail Settings page Configuration tab gt Advanced Applications menu gt Voice Mail Settings page item Figure 3 92 Voice Mail Settings Page Line Transfer Mode Voice Mail Interface Digit Patterns Forward on Busy Digit Pattern Internal Forward on Do Not Disturb Digit Pattern Internal Forward on No Reason Digit Pattern Internal Forward on No Answer Digit Pattern Internal Forward on Busy Digit Pattern External Forward on No Answer Digit Pattern External Forward on Do Not Disturb Digit Pattern External Forward on No Reason Digit Pattern External
370. ferent IP addresses each associated with a unique VLAN ID The configuration is performed using the Multiple Overview of Multiple Interface Table Interface table which is configurable using the ini file Web and SNMP interfaces The Multiple Interfaces scheme allows you to define up to 16 different IP addresses and VLANs in a table format as shown below Table 10 1 Multiple Interface Table Interface IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IPv4 IP Address 10 31 174 50 10 32 174 50 10 33 174 50 10 34 174 50 10 35 174 50 10 36 174 50 10 37 174 50 10 38 174 50 10 39 174 50 10 40 174 50 10 41 174 50 10 42 174 50 10 43 174 50 10 44 174 50 10 45 174 50 10 46 174 50 Prefix Length 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 Default Gateway 0 0 0 0 0 0 0 0 10 33 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 VLAN ID 14 15 16 17 18 19 Interface Name ManagementIF ControllF Media1IF Media2IF Media3IF Media4IF Media5IF Media6IF Media7IF Media8IF Media9IF Media10IF Media111F Media12IF Media131F Media14IF Complementing the network configuration are some VLAN related parameters determining if VLANs are enabled and the Native VLAN ID refer to the sub sections below as well as VLAN priorities and DiffServ val
371. figure up to nine Tel Profiles i e indices 1 through 9 The parameter IpPreference determines the priority of the Tel Profile 1 to 20 where 20 is the highest preference If both IP and Tel Profiles apply to the same call the coders and common parameters i e parameters configurable in both IP and Tel Profiles of the preferred profile are applied to that call If the Tel and IP Profiles are identical the Tel Profile parameters take precedence The parameter EnableVoiceMailDelay is applicable only if voice mail is enabled globally using the parameter VoiceMaillnterface To use the settings of the corresponding global parameter enter the value 1 288 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Fora detailed description of each parameter refer to its corresponding global parameter Fora description of using ini file table parameters refer to Configuring ini File Table Parameters on page 186 6 8 Supplementary Services Parameters This subsection describes the device s supplementary telephony services parameters 6 8 1 Caller ID Parameters The caller ID parameters are described in the table below Table 6 34 Caller ID Parameters Parameter Description Web Caller ID Permissions Table EMS SIP Endpoints gt Caller ID EnableCallerlD This ini file table parameter configures Caller ID permissions It allows you to enable or disable per
372. fix The prefix is separated by a comma from the number of additional digits m The prefix can include numerical ranges in the format x y as well as multiple numerical ranges n m x y no comma between them The prefix can include asterisks and number signs The number of additional digits can include a numerical range in the format x y m Empty lines and lines beginning with a semicolon are ignored Note If the external Dial Plan file is used for digit mapping rules then the parameter DigitMapping is ignored An example of a Dial Plan file with indices in ini file format before conversion to binary dat is shown below PLAN1 Area codes 02 03 phone numbers include 7 digits 02 7 03 7 Cellular VoIP area codes 052 054 phone numbers include 8 digits 052 8 054 8 International prefixes 00 012 014 number following prefixes include 7 to 14 digits 00 7 14 O12 7 14 014 7 14 Emergency number 911 no additional digits expected SIP User s Manual 380 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities Pill 0 PLAN2 Supplementary services such as Call Camping and Last Calls no additional digits expected by dialing 41 42 or 43 4 1 3 0 9 2 9 2 1 Routing Applications Stand Alone Survivability SAS Feature The device s Stand Alone Survivability SAS feature ensures telephony communicatio
373. for channel allocation for IP to Tel calls if the Hunt Group s Channel Select Mode is set to By Dest Phone Number Note If the this field includes alphabetical characters and the phone number is defined for a range of channels e g 1 4 then the phone number must end with a number e g user1 Hunt Group ID The Hunt Group ID 1 99 assigned to the corresponding channels The same Hunt Group ID can be assigned to more than one group of channels The Hunt Group ID is used to define a group of common channel behavior that are used for routing IP to Tel calls If an IP to Tel call is assigned to a Hunt Group the call is routed to the channel s pertaining to that Hunt Group ID Notes Once you have defined a Hunt Group you must configure the parameter PSTNPrefix IP to Hunt Group Routing Table to assign incoming IP calls to the appropriate Hunt Group If you do not configure this table calls cannot be established You can define the method for which calls are assigned to channels within the Hunt Groups using the parameter TrunkGroupSettings Tel Profile ID The Tel Profile ID assigned to the channels Note For configuring Tel Profiles refer to the parameter TelProfile 3 3 4 11 SAS Parameters The SAS submenu allows you to configure the SAS application This submenu includes the Stand Alone Survivability item page refer to Configuring Stand Alone Survivability Parameters on page 145 from which y
374. fset min wv Telnet Settings Embedded Telnet Server Disable r Telnet Server TCP Port 23 Telnet Server Idle Timeout 0 SSH Server Enable Disable SSH Server Port 22 v DNS Settings DNS Primary Server IP DNS Secondary Server IP v STUN Settings Enable STUN Disable STUN Server Primary IP 0 0 0 0 STUN Server Secondary IP w NFS Settings NFS Table uj w DHCP Settings Enable DHCP Disable 2 Configure the parameters as required For configuring NFS under the NFS Settings group click the NFS Table gt button the NFS Settings page appears For a description on configuring this page refer to Configuring the NFS Settings on page 56 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 55 March 2010 7a tall AudioCodes MediaPack Series 3 3 1 3 Configuring the NFS Settings Network File System NFS enables the device to access a remote server s shared files and directories and to handle them as if they re located locally You can configure up to 16 different NFS file systems As a file system the NFS is independent of machine types operating systems and network architectures NFS is used by the device to load the cmp ini and auxiliary files using the Automatic Update mechanism refer to t
375. g this global parameter Web Enable VoiceMail URI Enables or disables the interworking of target and cause EMS Enable VMURI for redirection from Tel to IP and vice versa according to EnableVMURI RFC 4468 0 Disable Disable default 1 Enable Enable Web EMS Line Transfer Mode Determines the call transfer method used by the device LineTransferMode 0 None IP default 1 Blind PBX blind transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX dials the digits that are received in the Refer To header and then immediately drops the line on hook The PBX performs the transfer internally 2 Semi Supervised PBX Semi Supervised transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX and then dials SIP User s Manual 274 Document LTRT 65413 SIP User s Manual Parameter SMDI Parameters Web EMS Enable SMDI SMDI Web EMS SMDI Timeout SMDITimeOut Version 6 0 6 Configuration Parameters Reference Description the digits that are received in the Refer To header If no Busy or Reorder tones are detected within approximately 2 seconds the device completes the call transfer by releasing the line otherwise the transfer is cancelled the device sends a SIP NOTIFY message with a failure reason in the NOTIFY body such as 486 if busy tone detected and generates an additional hook flash towards
376. ge 419 To activate these supplementary services enable each service s corresponding parameter using the Web interface or ini file All call participants must support the specific supplementary service that is used When working with certain application servers such as BroadSof s BroadWorks in client server mode the application server controls all supplementary services and keypad features by itself the device s supplementary services must be disabled 9 7 1 Call Hold and Retrieve Initiating Call Hold and Retrieve Version 6 0 Active calls can be put on hold by pressing the phone s hook flash button The party that initiates the hold is called the holding party the other party is called the held party After a successful Hold the holding party hears a Dial tone HELD_TONE defined in the device s Call Progress Tones file Call retrieve can be performed only by the holding party while the call is held and active The holding party performs the retrieve by pressing the telephone s hook flash button After a successful retrieve the voice is connected again Hold is performed by sending a Re INVITE message with IP address 0 0 0 0 or a sendonly in the SDP according to the parameter HoldFormat 409 March 2010 7a c tal AudioCodes MediaPack Series Receiving Hold Retrieve m When an active call receives a Re INVITE message with either the IP address 0 0 0 0 or the inactive string in SDP the device st
377. ge is 0 to 2 592 000 The default value is 30 Note For this parameter to take effect a device reset is required Global public IP address of the device to enable static NAT between the device and the Internet Note For this parameter to take effect a device reset is required Enables or disables the NAT mechanism 0 Enabled 1 Disabled default Note The compare operation that is performed on the IP address is enabled by default and is configured by the parameter EnablelPAddrTranslation The compare operation that is performed on the UDP port is disabled by default and is configured by the parameter EnableUDPPortTranslation 215 March 2010 A K tal AudioCodes MediaPack Series Parameter Description EnablelPAddrTranslation Enables IP address translation for RTP RTCP and T 38 packets 0 Disable IP address translation 1 Enable IP address translation default 2 Enable IP address translation for RTP Multiplexing ThroughPacket 3 Enable IP address translation for all protocols RTP RTCP T 38 and RTP Multiplexing When enabled the device compares the source IP address of the first incoming packet to the remote IP address stated in the opening of the channel If the two IP addresses don t match the NAT mechanism is activated Conseguently the remote IP address of the outgoing stream is replaced by the source IP address of the first incoming packet Notes The N
378. ge is received the call is established The valid range is 0 to 10000 i e 10 seconds The default value is 2000 275 March 2010 ca AudioCodes Parameter MediaPack Series Description Message Waiting Indication MWI Parameters Web MWI Off Digit Pattern EMS MWI Off Code MWIOffCode Web MWI On Digit Pattern EMS MWI On Code MWIOnCode Web MWI Suffix Pattern EMS MWI Suffix Code MWISuffixCode Web MWI Source Number EMS MWI Source Name MWISourceNumber Determines the digit code used by the device to notify the PBX that there aren t any messages waiting for a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string Determines the digit code used by the device to notify the PBX of messages waiting for a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string Determines the digit code used by the device as a suffix for MWI On Digit Pattern and MWI Off Digit Pattern This suffix is added to the generated DTMF string after the extension number The valid range is a 25 character string Determines the calling party s phone number used in the Q 931 MWI Setup message to PSTN If not configured the channel s phone number is used as the calling number Digit Patterns The following digit pattern parameters apply only to voice mail applications that use the DTMF communication method For
379. gement tab on the Navigation bar displays menus in the Navigation tree related to device management These menus include the following m Management Configuration refer to Management Configuration on page 152 m Software Update refer to Software Update on page 163 Management Configuration The Management Configuration menu allows you to configure the device s management parameters This menu contains the following page items m Management Settings refer to Configuring the Management Settings on page 152 m Regional Settings refer to Configuring the Regional Settings on page 158 m Maintenance Actions refer to Maintenance Actions on page 159 Configuring the Management Settings The Management Settings page allows you to configure the device s management parameters For detailed description on the SNMP parameters refer to SNMP Parameters on page 242 SIP User s Manual 152 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To configure the management parameters 1 Open the Management Settings page Management tab gt Management Configuration menu gt Management Settings page item Figure 3 95 Management Settings Page v Syslog Settings Enable Syslog Disable Syslog Server IP Address Syslog Server Port 514 Debug Level 0 4nalog Ports Filter 1 w SNMP Settings SNMP Trap Destinations SNMP Community String SNMP V3 Table SNMP Trusted M
380. guests are sent is the IP address defined for the Proxy Set ID refer to Configuring the Proxy Sets Table on page 97 associated with this IP Group This occurs only in the following conditions The parameter Registration Mode is set to Per Account in the Hunt Group Settings table refer to Configuring Hunt Group Settings on page 85 The parameter Register in this table is set to 1 In addition for a SIP call that is identified by both the Served Hunt Group and Serving IP Group the username and password for digest authentication defined in this table is used For Tel to IP calls the Serving IP Group is the destination IP Group defined in the Hunt Group Settings table or Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 For IP to Tel calls the Serving IP Group is the Source IP Group ID defined in the IP to Hunt Group Routing Table refer to Configuring the IP to Hunt Group Routing Table on page 131 Note If no match is found in this table for incoming or outgoing calls the username and password defined in the Authentication table refer to Configuring Authentication on page 136 or the global parameters UserName and Password defined on the Proxy amp Registration page Username Digest MD5 Authentication user name up to 50 characters Account_Username SIP User s Manual 94 Document LTRT 65413 SIP User s Manual Parameter Password Account_Password Ho
381. guration Parameters on page 245 For defining groups of coders which can be assigned to Tel and IP Profiles refer to Configuring Coder Groups on page 104 The device always uses the packetization time requested by the remote side for sending RTP packets For an explanation on V 152 support and implementation of T 38 and VBD coders refer to Supporting V 152 Implementation on page 408 gt To configure the device s coders 1 Open the Coders page Configuration tab gt Protocol Configuration menu gt Coders And Profile Definitions submenu gt Coders page item Figure 3 64 Coders Page Packetization Payload Silence Coder Name Tne Type Suppression Disabled SIP User s Manual 102 Document LTRT 65413 SIP User s Manual 3 Web Based Management 2 From the Coder Name drop down list select the required coder 3 From the Packetization Time drop down list select the packetization time in msec for the selected coder The packetization time determines how many coder payloads are combined into a single RTP packet 4 From the Rate drop down list select the bit rate in kbps for the selected coder 5 Inthe Payload Type field if the payload type i e format of the RTP payload for the selected coder is dynamic enter a value from 0 to 120 payload types of well known coders cannot be modified 6 From the Silence Suppression drop down lis
382. guration Parameters Reference on page 207 Version 6 0 109 March 2010 ca AudioCodes MediaPack Series gt To configure the advanced general protocol parameters 1 2 3 4 Open the Advanced Parameters page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Advanced Parameters page item Figure 3 68 Advanced Parameters Page wv General IP Security Filter Calls to IP Enable Digit Delivery to Tel jd Enable Digit Delivery to IP Enable DID Wink Delay Before DID Wink Reanswer Time PSTN Alert Timeout Disable Dont Filter Disable Disable Disable 0 0 180 wv Disconnect and Answer Supervision Send Digit Pattern on Connect Enable Polarity Reversal Enable Current Disconnect Disconnect on Broken Connection Broken Connection Timeout 100 msec Disconnect Call on Silence Detection i Silence Detection Period sec Silence Detection Method Enable Fax Re Routing Disable Disable Yes 100 No 120 Voice Energy Detectors Disable w CDR and Debug CDR Server IP Address CDR Report Level Debug Level w Misc Parameters Progress Indicator to IP Enable Busy Out Graceful Busy Out Timeout sec Default Release Cause Max Number of Active Calls Max Call Duration min i Enable LAN W
383. harge Code 100 10 33 45 63 Not Configurea wi Not Configured w 1 1 0 30 40 i 10 33 45 64 Not Configured v v o 5 7 9 I domain com Not Configured v lo x lo 00 Nat Configured v The figure above shows the following configured Tel to IP routing rules e Rule 1 If the called phone prefix is 10 and the caller s phone prefix is 100 the call is assigned settings configured for IP Profile ID 1 and sent to IP address 10 33 45 63 e Rule 2 If the called phone prefix is 20 and the caller is all prefixes the call is sent to the destination according to IP Group 1 which in turn is associated with a Proxy Set ID providing the IP address e Rule 3 If the called phone prefix is between 30 and 40 and the caller belongs to Hunt Group ID 1 the call is sent to IP address 10 33 45 64 e Rule 4 If the called phone prefix is either 5 7 8 or 9 and the caller is all the call is sent to domain com e Rule 5 If the called phone prefix is 00 and the caller is all the call is discarded 2 From the Routing Index drop down list select the range of entries that you want to add Configure the Tel to IP routing rules according to the table below 4 Click the Submit button to apply your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 128 Document LTRT 65413 SIP User s Manual Para
384. he Product Reference Manual Note that an NFS file server can share multiple file systems There must be a separate row for each remote file system shared by the NFS file server that needs to be accessed by the device gt Toadd remote NFS file systems 1 Open the Application Settings page refer to Configuring the Application Settings on page 54 2 Under the NFS Settings group click the NFS Table p button the NFS Settings page appears Figure 3 37 NFS Settings Page Authentication Type Index Host Or IP Root Path NFS Version User ID vlan Type l1 101345 i audiofiles NFSVersion3 M h Jo 1 moa v 3 In the Ad field enter the index number of the remote NFS file system and then click Add an empty entry row appears in the table 4 Configure the NFS parameters according to the table below 5 Click the Apply button the remote NFS file system is immediately applied which can be verified by the appearance of the NFS mount was successful message in the Syslog server 6 To save the changes to flash memory refer to Saving Configuration on page 161 To avoid terminating current calls a row must not be deleted or modified while the device is currently accessing files on that remote NFS file system The combination of Host Or IP and Root Path must be unique for each row in the table For example the table must include only one row with a Host IP of 192 168 1 1
385. he Retry After SIP header in SIP 503 Service Unavailable responses to indicate an unavailable service The Retry After header is used with the 503 Service Unavailable response to indicate how long the service is expected to be unavailable to the reguesting SIP client The device maintains a list of available proxies by using the Keep Alive mechanism The device checks the availability of proxies by sending SIP OPTIONS every keep alive timeout to all proxies If the device receives a SIP 503 response to an INVITE it also marks that the proxy is out of service for the defined Retry After period Determines the device usage of the P Associated URI header This header can be received in 200 OK responses to REGISTER requests When enabled the first URI in the P Associated URI header is used in subsequent requests as the From P Asserted Identity headers value 0 Disable default 1 Enable Note P Associated URIs in registration responses is handled only if the device is registered per endpoint Determines the SIP header used for the source number in incoming INVITE messages empty string Use the device s internal logic for header preference default The logic for filling the calling party parameters is as follows the SIP header is selected first from which the calling party parameters are obtained first priority is P Asserted Identity second is Remote Party ID and third is the From header Once a URL is
386. he Tel side Automatic Dialing Automatic dialing is defined using the ini file parameter table TargetOfChannel refer to Analog Telephony Parameters or the embedded Web server s Automatic Dialing screen refer to Automatic Dialing on page 137 The SIP call flow diagram below illustrates Automatic Dialing Figure 9 6 Call Flow for Automatic Dialing SIP Client F1 INVITE Sent immediately if Caller ID detected otherwise sent after 2 rings or after 1 ring if RingsBeforeCallerlD 0 FXO Gateway FXO seizes line off hook only after receiving 200 OK even after receiving 183 to enable routing to voice mail on the PBX side SIP User s Manual 390 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 4 2 2 2 Collecting Digits Mode When automatic dialing is not defined the device collects the digits The SIP call flow diagram below illustrates the Collecting Digits Mode Figure 9 7 Call Flow for Collecting Digits Mode SIP Client FXO Gateway FXO detects ring on line FXO detects Caller ID according to RingsBeforeCallerlD F1 INVITE Sent after collecting MaxDigits or after TimeBetweenDigits has expired or once digit strings DigitMapping match digit map 5 9 4 2 2 3 FXO Supplementary Services The FXO supplementary services include the following m Hold Transfer toward the Tel side The ini file parameter LineTransferMode must be set to 0 default If the FXO receives a
387. he address is configured on the Multiple Interface Table page refer to Configuring the Multiple Interface Table on page 50 The maximum number of times a packet can be forwarded hops between the device and destination typically up to 20 Note This parameter must be set to a number greater than 0 for the routing rule to be valid Routing entries with Hop Count eguals 0 are local routes set automatically by the device Specifies the interface network type to which the routing rule is applied 0 OAMP default 1 Media 2 Control For detailed information on the network types refer to Configuring the Multiple Interface Table on page 50 59 March 2010 7a e AudioCodes MediaPack Series 3 3 1 5 Configuring the QoS Settings The QoS Settings page is used for configuring the Quality of Service QoS parameters This page allows you to assign VLAN priorities IEEE 802 1p and Differentiated Services DiffServ for the supported Class of Service CoS For a detailed description of the parameters appearing on this page refer to Networking Parameters on page 207 For detailed information on IP QoS using DiffServ refer to IP QoS via Differentiated Services DiffServ on page 448 gt To configure QoS 1 Open the QoS Settings page Configuration tab gt Network Settings menu gt QoS Settings page item Figure 3 39 QoS Settings Page v Priority Settings Network Pri
388. he default is 40 353 March 2010 ca AudioCodes Parameter EMS Enable Inband Network Detection EnableFaxModemInbandNetwork Detection EMS NSE Mode NSEMode EMS NSE Payload Type NSEPayloadType Web V 21 Modem Transport Type EMS V21 Transport V21ModemTransportType SIP User s Manual MediaPack Series Description Enables or disables in band network detection related to fax modem 0 Disable default 1 Enable When this parameter is enabled on Bypass and transparent with events mode VxxTransportType 2 or 3 a detection of an Answer Tone from the network triggers a switch to bypass mode in addition to the local Fax Modem tone detections However only a high bit rate coder voice session effectively detects the Answer Tone sent by a remote endpoint This can be useful when for example the payload of voice and bypass is the same allowing the originator to switch to bypass mode as well Cisco compatible fax and modem bypass mode 0 NSE disabled default 1 NSE enabled Notes This feature can be used only if VxxModemTransportType 2 Bypass If NSE mode is enabled the SDP contains the following line a rtpmap 100 X NSE 8000 To use this feature v The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw v Set the Modem transport type to Bypass mode VxxModemTransportT ype 2 for all modems v Configure
389. he destination number of Tel to IP calls The format of this parameter is as follows NumberMapTel2Ip FORMAT NumberMapTel2Ip Index NumberMapTel2Ip DestinationPrefix NumberMapTel2Ip SourcePrefix NumberMapTel2Ip SourceAddress NumberMapTel2Ip NumberType NumberMapTel2Ip NumberPlan NumberMapTel2Ip RemoveFromLeft NumberMapTel2Ip RemoveFromRight NumberMapTel2Ip LeaveFromRight NumberMapTel2Ip Prefix2Add NumberMapTel2Ip Suffix2Add NumberMapTel2Ip IsPresentationRestricted NumberMapTel2lp_SrcTrunkGroupID NumberMapTel2Ip _ SrclPGroupID NumberMapTel2Ip For example NumberMapTel2Ip 0 01 0 0 2 5 971 NumberMapTel2Ip 1 10 10 255 255 3 0 5 100 255 Notes This table parameter can include up to 120 indices 0 119 The parameters SourceAddress and IsPresentationRestricted are not applicable The parameters SrclPGroupID NumberType and NumberPlan are not applicable The parameters RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add and LeaveFromRight are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and then Suffix2Add To configure manipulation of destination numbers for Version 6 0 345 March 2010 ca AudioCodes Parameter MediaPack Series Description Tel to IP calls using the
390. he device endpoints and SIP Accounts are registered to a SIP Registrar Proxy server gt To view Registration status m Open the Registration Status page Status amp Diagnostics tab gt Gateway Statistics menu gt Registration Status page item Figure 3 117 Registration Status Page Registered Per Gateway wv Ports Registration Status Gateway Port Status Port 1 FXS NOT REGISTERED Port 2 FXS NOT REGISTERED Port 3 FXS Port4 FXS NOT REGISTERED Port5 FXO NOT REGISTERED Port 6 FXO NOT REGISTERED Port 7 FXO NOT REGISTERED Port 8 FRO NOT REGISTERED wv Accounts Registration Status Index Group Type Group Name Status 1 Trunk Group NOT REGISTERED 2 NA 3 NA 4 NA m Registered Per Gateway e YES registration is per device e NO registration is not per device m Ports Registration Status e REGISTERED channel is registered e NOT REGISTERED channel not registered m Accounts Registration Status registration status based on the Accounts table configured in Configuring the Account Table on page 93 e Group Type type of served group Hunt Group or IP Group e Group Name name of the served group if applicable e Status indicates whether or not the group is registered Registered or Unregistered The registration mode i e per device endpoint account or no registration is configured in the Hunt Group Settings
391. he device to global IP numbers In this context a global IP phone number alphanumerical serves as a routing identifier for calls in the IP world The PBX extension uses this mapping to emulate the behavior of an IP phone Note By default the mapping mechanism is disabled and must be activated using the parameter EnableUserlnfoUsage The maximum size of the file is 10 800 bytes Each line in the file represents a mapping rule of a single PBX extension Up to 100 rules can be configured Each line includes five items separated with commas The items are described in the table below Table 8 1 User Information Items Item Description Maximum Size Characters PBX extension The relevant PBX extension number 10 Global phone The relevant global phone number 20 A string that represents the PBX extensions for the Caller ID 30 Display name SIP User s Manual 374 Document LTRT 65413 SIP User s Manual 8 Auxiliary Configuration Files Item Description Maximum Size Characters A string that represents the user name for SIP usermame registration 40 Password A string that represents the password for SIP 20 registration For FXS ports when the device is required to send a new request with the Authorization header for example after receiving a SIP 401 reply it uses the user name and password from the Authentication table To use the username and password from the User Info file change the parameter Pa
392. he device to operate with SNMPv3 via EMS to a non configured System In the MG Tree select the required Region to which the device belongs the device is displayed in the Main pane Right click the device and then from the shortcut menu point to Configuration and then click SNMP Configuration the SNMP Configuration window appears Figure 5 9 SNMP Configuration Screen SNMP Configuration J SNMPv2 SNMPv3 SNMP Engine ID Security Name snmpv3user Security Level Authentication amp Privacy Authentication Protocol SHA Authentication Key a Privacy Protocol AES 128 W Update Media Gateway SNMP Settings OK Cancel Select the SNMPv3 option Configure the SNMPv3 fields and then select the Update Media Gateway SNMP Settings check box Click OK the update progress is displayed Click Done when complete Open the SNMPv3 Users screen Configuration icon gt Network Frame menu gt SNMPv3 Users tab From the SNMPv3 Users tab s drop down list choose Unit value the SNMPv3 Users table is refreshed with the values that you entered in Step 4 Click the Save button the EMS and the device are now synchronized 203 March 2010 A tal AudioCodes MediaPack Series 5 8 4 5 9 Cloning SNMPv3 Users According to the SNMPv3 standard SNMPv3 users on the SNMP Agent on the device cannot be added via the SNMP protocol e g SNMP Manager i e the EMS Instead new users mus
393. he last trap manager entry of snmpTargetAddrTable in the snmpTargetMIB For example mngr corp mycompany com The valid range is a 99 character string 243 March 2010 ca AudioCodes Parameter SNMP Community String Parameters Community String SNMPReadOnlyCommunityString_x Community String SNMPReadWriteCommunityString_x Trap Community String SNMPTrapCommunityString Web SNMP V3 Table EMS SNMP V3 Users SNMPUsers SIP User s Manual MediaPack Series Description Defines up to five read only SNMP community strings up to 19 characters each The default string is public Defines up to five read write SNMP community strings up to 19 characters each The default string is private Community string used in traps up to 19 characters The default string is trapuser This ini file table parameter configures SNMP v3 users The format of this parameter is as follows SNMPUsers FORMAT SNMPUsers Index SNMPUsers Username SNMPUsers AuthProtocol SNMPUsers PrivProtocol SNMPUsers AuthKey SNMPUsers PrivKey SNMPUsers Group SNMP Users For example SNMPUsers 1 v3admin1 1 0 myauthkey 1 The example above configures user v3admin1 with security level authNoPriv 2 authentication protocol MD5 authentication text password myauthkey and ReadWriteGroup2 Notes This parameter can include up to 10 indices Fora description of this table s individual parameters a
394. he parameter FXOBetweenRingTime the FXO device doesn t initiate a call to the IP m Automatic dialing is enabled if the remote party doesn t answer the call and the ringing signal stops for a user defined time using the parameter FXOBetweenRingTime the FXO device releases the IP call Ring Detection Timeout supports full ring cycle of ring on and ring off from ring start to ring start Version 6 0 393 March 2010 tall AudioCodes MediaPack Series 9 4 3 Remote PBX Extension Between FXO and FXS Devices Remote PBX extension offers a company the capability of extending the power of its local PBX by allowing remote phones remote offices to connect to the company s PBX over the IP network instead of via PSTN This is as if the remote office is located in the head office where the PBX is installed PBX extensions are connected through FXO ports to the IP network instead of being connected to individual telephone stations At the remote office FXS units connect analog phones to the same IP network To produce full transparency each FXO port is mapped to an FXS port i e one to one mapping This allows individual extensions to be extended to remote locations To call a remote office worker a PBX user or a PSTN caller simply dials the PBX extension that is mapped to the remote FXS port This section provides an example on how to implement a remote telephone extension through the IP network using 8 port FXO and 8 port FX
395. he right pointing arrow the pane is displayed and the button is replaced by the left pointing arrow button Figure 3 6 Showing and Hiding Navigation Pane Show Hide gt Button Displayed Navigation QoS Settings Delete Selected Ermes ti Pane ta i ume Semegs Destrepen IP Address Oewnabon Mask Jaewey IP Address Hep Count Add New Entry Aad New Eray 3 1 6 Working with Configuration Pages The configuration pages contain the parameters for configuring the device The configuration pages are displayed in the Work pane which is located to the right of the Navigation pane Version 6 0 29 March 2010 K tal AudioCodes MediaPack Series 3 1 6 1 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation tree gt To open a configuration page in the Work pane 1 On the Navigation bar click the required tab e Configuration refer to Configuration Tab on page 50 e Management refer to Management Tab on page 151 e Status amp Diagnostics refer to Status amp Diagnostics Tab on page 172 The menus of the selected tab appears in the Navigation tree 2 In the Navigation tree drill down to the required page item the page opens in the Work pane You can also access previously opened pages by clicking your Web browser s Back button until you have reached the required page This is useful if you want to view pages in which you have performed configur
396. he transfer service is enabled the user can activate Transfer using hook flash signaling If this service is enabled the remote party performs the call transfer Notes To use call transfer the devices at both ends must support this option To use call transfer set the parameter EnableHold to 1 Defines the string that is added as a prefix to the transferred forwarded called number when the REFER 3xx message is received Notes The number manipulation rules apply to the user part of the Refer To and or Contact URI before it is sent in the INVITE message This parameter can be used to apply different manipulation rules to differentiate transferred forwarded number from the originally dialed number Defines the prefix that is added to the destination number received in the SIP Refer To header for IP to Tel calls This parameter is applicable to FXO Blind Transfer modes LineTransferMode 1 2 or 3 The valid range is a string of up to 9 characters The default is an empty string Determines the device behavior when Transfer is initiated while in Alerting state 0 Disable Send REFER with the Replaces header default 1 Enable Send CANCEL and after a 487 response is received send REFER without the Replaces header Determines whether the device adds the Blind Transfer code KeyBlindTransfer to the dialed destination number 0 Disable default 1 Enable 302 Document LTRT 65413
397. hen the server cannot be Response contacted OCSPDefaultResponse 0 Rejects peer certificate default 1 Allows peer certificate 6 5 RADIUS Parameters The RADIUS parameters are described in the table below For detailed information on the supported RADIUS attributes refer to Supported RADIUS Attributes on page 436 Parameter Web Enable RADIUS Access Control EnableRADIUS Web Accounting Server IP Address RADIUSAccServerIP Web Accounting Port RADIUSAccPort Web EMS RADIUS Accounting Type RADIUSAccountingType Web AAA Indications EMS Indications AAAlndications Web Device Behavior Upon RADIUS Timeout BehaviorUponRadiusTimeout MaxRADIUSSessions RADIUSRetransmission SIP User s Manual Table 6 25 RADIUS Parameters Description Determines whether the RADIUS application is enabled 0 Disable RADIUS application is disabled default 1 Enable RADIUS application is enabled Note For this parameter to take effect a device reset is reguired IP address of the RADIUS accounting server Port of the RADIUS accounting server The default value is 1646 Determines when the RADIUS accounting messages are sent to the RADIUS accounting server 0 At Call Release Sent at call release only default 1 At Connect amp Release Sent at call connect and release 2 At Setup 8 Release Sent at call setup and release Determines the Authentication A
398. hernet Port Information on page 173 IP Interface Status refer to Viewing Active IP Interfaces on page 174 Device Information refer to Viewing Device Information on page 174 Performance Statistics refer to Viewing Performance Statistics on page 175 Active Alarms refer to Viewing Active Alarms on page 176 Viewing the Device s Syslog Messages The Message Log page displays Syslog debug messages sent by the device You can select the Syslog messages in this page and then copy and paste them into a text editor such as Notepad This text file txt can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting Note It s not recommended to keep a Message Log session open for a prolonged period This may cause the device to overload For prolonged and detailed debugging use an external Syslog server refer to the Product Reference Manual SIP User s Manual 172 Document LTRT 65413 SIP User s Manual 3 Web Based Management To activate the Message Log Set the parameter Debug Level GwDebugLevel to 7 refer Configuring Advanced Parameter on page 109 This parameter determines the Syslog logging level in the range 0 to 6 where 7 is the highest level Open the Message Log page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Message Log page item the Message Log page is displayed and the log is activated Figure 3 110 Message Log Screen Version 6 0
399. hhook Notes FXO interfaces support only the receipt of RFC 2833 Hook Flash signals and INFO 1 type FXS interfaces send Hook Flash signals only if the parameter EnableHold is set to 0 Defines the minimum time in msec for detection of a hook flash event Detection is guaranteed for hook flash periods of at least 60 msec when setting the minimum time to 25 Hook flash signals that last a shorter period of time are ignored The valid range is 25 to 300 The default value is 300 Notes For this parameter to take effect a device reset is required This parameter is applicable only to FXS interfaces It s recommended to reduce the detection time by 50 msec from the desired value For example if you want to set the value to 200 msec then enter 150 msec i e 200 minus 50 279 March 2010 ca AudioCodes Parameter Web Max Flash Hook Detection Period msec EMS Flash Hook Period FlashHookPeriod DTMF Parameters EMS Use End of DTMF MGCPDTMFDetectionPoint Web Declare RFC 2833 in SDP EMS Rx DTMF Option RxDTMFOption Web EMS Tx DTMF Option TxDTMFOption SIP User s Manual MediaPack Series Description Defines the hook flash period in msec for both Tel and IP sides per device For the IP side it defines the hook flash period that is reported to the IP For the analog side it defines the following FXS interfaces v Maximum hook flash detection period A longer signal
400. hird Proposal Authentication Type Not Defined Third Proposal OH Group Not Defined Fourth Proposal Encryption Type Not Defined Fourth Proposal Authentication Type Not Defined Server Certificate File Loaded Trusted Root File Loaded Apply Policy Index O State Este Authentication Method Pre zhared Key Shered Key IKE SA LifeTime sec 28800 IKE SA LifeTime KB 0 SETETE E SETE SE Click the Apply button to load the certificates future IKE negotiations are now performed using the new certificates 3 3 3 4 2 Client Certificates By default Web servers using SSL provide one way authentication The client is certain that the information provided by the Web server is authentic When an organizational PKI is used two way authentication may be desired both client and server should be authenticated using X 509 certificates This is achieved by installing a client certificate on the managing PC and loading the same certificate in base64 encoded X 509 format to the device s Trusted Root Certificate Store The Trusted Root Certificate file should contain both the certificate of the authorized user and the certificate of the CA Since X 509 certificates have an expiration date and time the device must be configured to use NTP refer to Simple Network Time Protocol Support on page 447 to obtain the current date and time Without the correct date and time client certificates cannot work gt 1 Version 6 0 To enable two w
401. his mode set the parameter FaxCNGMode to 1 2 Events Only CNG is detected on the originating side and a fax session is started by the originating side using the Re INVITE message Usually T 38 fax session starts when the preamble signal is detected by the answering side Some SIP devices don t support the detection of this fax signal on the answering side and thus in these cases it is possible to configure the device to start the T 38 fax session when the CNG tone is detected by the originating side However this mode is not recommended Number of 20 msec coder payloads that are used to generate a fax modem bypass packet The valid range is 1 2 or 3 coder payloads The default value is 1 coder payload 352 Document LTRT 65413 SIP User s Manual Parameter FaxModemNTEMode Web EMS Fax Bypass Payload Type FaxBypassPayloadType EMS Modem Bypass Payload Type ModemBypassPayloadType EMS Relay Volume dBm FaxModemRelayVolume Web EMS Fax Bypass Output Gain FaxBypassOutputGain Web EMS Modem Bypass Output Gain ModemBypassOutputGain EMS NTE Max Duration NTEMaxDuration EMS Basic Packet Interval FaxModemBypassBasicRTPPack etinterval EMS Dynamic Jitter Buffer Minimal Delay dB FaxModemBypassDJBufMinDela y Version 6 0 6 Configuration Parameters Reference Description Determines whether the device sends RFC 2833 ANS ANSam events upon detection of fax and or mode
402. hone connected to port 2 of the same device Listen for progress tones at the calling phone and for the ringing tone at the called phone Answer the called phone speak into the calling phone and check the voice quality Dial 201 from the phone connected to port 1 of the first device the phone connected to port 1 of the second device rings Answer the call and check the voice quality 427 March 2010 ra z a AudioCodes MediaPack Series 9 8 5 SIP Trunking between Enterprise and ITSPs By implementing the device s enhanced and flexible routing capabilities you can design complex routing schemes This section provides an example of an elaborate routing scheme for SIP trunking between an Enterprise and two Internet Telephony Service Providers ITSP using AudioCodes device Scenario In this example an Enterprise has deployed the device with eight FXS interfaces The first four phones operate with ITSP 1 using UDP while the next four phones channels 5 8 operate with ITSP 2 using TCP ITSP 1 requires single registration i e one registration for all four phones while ITSP 2 requires registration per phone Each ITSP implements two servers for redundancy and load balancing The figure below illustrates this example setup Figure 9 23 Example Setup for Routing Between ITSPs and Enterprise PSTN Network Proxy Set 1 P 10 33 37 77 IP 10 33 37 79 ITSP 1 IP Group 1 POTS Phones j Hunt Group ID 1
403. hook flash the holding party hears a dial tone and can then initiate a new call which is called a Consultation call m While hearing a dial tone or when dialing to the new destination before dialing is complete the user can retrieve the held call by pressing hook flash Version 6 0 411 March 2010 A c tal AudioCodes MediaPack Series The held call can t be retrieved while Ringback tone is heard After the Consultation call is connected the user can toggle between the held and active call by pressing the hook flash key Note The Consultation feature is applicable only to FXS interfaces 9 7 4 Call Transfer There are two types of call transfers Consultation Transfer REFER and REPLACES The common method to perform a consultation transfer is as follows In the transfer scenario there are three parties Party A transferring Party B transferred Party C transferred to 1 A Calls B 2 B answers 3 A presses the hook flash button and places B on hold party B hears a hold tone 4 Adials C 5 After A completes dialing C A can perform the transfer by on hooking the A phone 6 After the transfer is complete B and C parties are engaged in a call The transfer can be initiated at any of the following stages of the call between A and C e Just after completing dialing C phone number transfer from setup e While hearing Ringback transfer from alert e While speaking to C transfer from acti
404. hooked TargetOfChannel 0 911 1 phone number 1002 is automatically dialed for Port 1 Notes This is parameter is applicable to FXS and FXO interfaces The indexing of this ini file table parameter starts at 0 Define this parameter for each device port that implements Automatic Dialing This parameter can appear up to 8 times for MP 118 port and up to 24 times for MP 124 devices After a ring signal is detected on an Enabled FXO port the device initiates a call to the destination number without seizing the line The line is seized only after the call is answered After a ring signal is detected on a Disabled or Hotline FXO port the device seizes the line For configuring this table using the Web interface refer to Configuring Automatic Dialing on page 137 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 306 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 8 11 Direct Inward Dialing Parameters The Direct Inward Dialing DID parameters are described in the table below Parameter Web EMS Enable DID Wink EnableDIDWink Web EMS Delay Before DID Wink DelayBeforeDIDWink EMS NTT DID Signalling Form NTTDIDSignallingForm EMS Enable DID EnableDID Version 6 0 Table 6 44 DID Parameters Description Enables Direct Inward Dialing DID using Wink Start signaling 0 Di
405. ia an SNMP MIB object refer to the Product Reference Manual If required the clock update is performed by the client as the final step of the update process The update is performed in such a way as to be transparent to the end users For instance the response of the server may indicate that the clock is running too fast on the client The client slowly robs bits from the clock counter to update the clock to the correct time If the clock is running too slow then in an effort to catch the clock up bits are added to the counter causing the clock to update quicker and catch up to the correct time The advantage of this method is that it does not introduce any disparity in the system time that is noticeable to an end user or that could corrupt call timeouts and timestamps Version 6 0 447 March 2010 A e AudioCodes MediaPack Series 10 7 10 8 10 8 1 IP QoS via Differentiated Services DiffServ DiffServ is an architecture providing different types or levels of service for IP traffic DiffServ according to RFC 2474 offers the capability to prioritize certain traffic types depending on their priority thereby accomplishing a higher level QoS at the expense of other traffic types By prioritizing packets DiffServ routers can minimize transmission delays for time sensitive packets such as VoIP packets The device can be configured to set a different DiffServ value to IP packets according to their class of service Network Pr
406. ications Enabling PAGE osudu dik kud ua ka lk b ldd Kubko x koks navy OF Hunt Group Settings Page a 85 SIP General Parameters Page c cccccccccsccssecsccsecssecseesecsacceesesecseecsecsecsaccsteseesssesssseessees OD DTMF DIN P KN IP Group Table Page 2 r 7 i Account Table Page Proxy amp Registration Page NE APNEA PE NE PAN EET AI dra eee Aloha r o AEST Proxy Sets Table ee EAE E AIAN IA PAE AA ERATI plane oe Coders Page pane 2 Coder Group Se IP Profile Sett ngs Page Advanced Parameters Pag Metering Tones Page Charge Codes Table Page Source Phon Number oo Table for Tel to IP Calls Redirect Number Tel to IP Page Phone Context Table Page Reasons for Alternative Routing Pag Routing General Parameters eases Tel to IP Routing Page Inbound IP Routing Table P ge Internal DNS Table Page Internal SRV Table Page PPOR TAS PROV O TS P POPP SO PP O POPP P Forward on Busy Trunk Des ination Page 136 Authentication Page A Automatic Dialing Page Caller Display Information Page p a boa E A tao n Call Forward Table Page 140 aller ID Permissions M i o P a JEL Call Waiting Page satan ee Endpoint Phone Number rTa le age SAS Configuration Page Voice Mail Settings Page is E E E E E AE E A E E E E oh RADIUS Parameters S NRS R eee seca OA er OVO 150 FXO Settings Page Management Setting Pag
407. ice reset is reguired 224 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 3 Debugging and Diagnostics Parameters This subsection describes the device s debugging and diagnostic parameters 6 3 1 General Parameters The general debugging and diagnostic parameters are described in the table below Table 6 13 General Debugging and Diagnostic Parameters Parameter EMS Enable Diagnostics EnableDiagnostics Web Enable LAN Watchdog EnableLanWatchDog WatchDogStatus LifeLineType Version 6 0 Description Checks the correct functionality of the different hardware components on the device On completion of the check and if the test fails the device sends information on the test results of each hardware component to the Syslog server 0 Rapid and Enhanced self test mode default 1 Detailed self test mode full test of DSPs PCM Switch LAN PHY and Flash 2 A quicker version of the Detailed self test mode full test of DSPs PCM Switch LAN PHY but partial test of Flash For detailed information refer to the Product Reference Manual Note For this parameter to take effect a device reset is reguired Determines whether the LAN Watch Dog feature is enabled 0 Disable Disable LAN Watch Dog default 1 Enable Enable LAN Watch Dog When LAN Watch Dog is enabled the device s overall communication integrity is checked periodically If
408. ice s behavior regarding call identifiers when a 3xx response is received for an outgoing INVITE request The device can either use the same call identifiers Call ID Branch To and From tags or change them in the new initiated INVITE 0 Forward Use different call identifiers for a redirected INVITE message default 1 Redirect Use the same call identifiers Enables the inclusion of the P Charging Vector header to all outgoing INVITE messages 0 Disable default 1 Enable 253 March 2010 ca AudioCodes Parameter Web EMS Retry After Time RetryAfterTime Web EMS Fake Retry After sec FakeRetryAfter Web EMS Enable P Associated URI Header EnablePAssociatedURIHeader Web EMS Source Number Preference SourceNumberPreference SelectSourceHeaderForCalled Number SIP User s Manual MediaPack Series Description Determines the time in seconds used in the Retry After header when a 503 Service Unavailable response is generated by the device The time range is 0 to 3 600 The default value is 0 Determines whether the device upon receipt of a SIP 503 response without a Retry After header behaves as if the 503 response included a Retry After header and with the period in seconds specified by this parameter 0 Disable Any positive value in seconds for defining the period When enabled this feature allows the device to operate with Proxy servers that do not include t
409. id 1 nas ip address 212 179 22 213 nas port type 0 acct status type 2 acct input octets acct output octets acct session time 1 4841 8800 acct input packets 122 acct output packets 220 called station id 201 calling station id 202 Accounting non standard parameters 4923 33 h323 gw id 4923 23 h323 remote address 212 179 22 214 4923 1 h323 ivr out h323 incoming conf id 02102944 600a1899 3 d61009 0e2f3cc5 4923 30 h323 disconnect cause 22 0x16 4923 27 h323 call type VOIP 4923 26 h323 call origin Originate 4923 24 h323 conf id 02102944 600a1899 3fd61009 Oe2f3cc5 9 13 SIP User s Manual Call Detail Record The Call Detail Record CDR contains vital statistic information on calls made by the device CDRs are generated at the end and optionally at the beginning of each call determined by the parameter CDRReportLevel and then sent to a Syslog server The destination IP address for CDR logs is determined by the parameter CDRSyslogServerIP For CDR in RADIUS format refer to Supported RADIUS Attributes on page 436 438 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities The following table lists the supported CDR fields Field Name ReportType Cid Callld Trunk BChan Conld TG EPTyp Orig Sourcelp Destlp TON NPI SrcPhoneNum SrcNumBeforeMap TON NPI DstPhoneNum DstNumBeforeMap Durat Coder Intrv Rtplp
410. id range is 0 to 32767 The default value is 100 Defines the Voice Silence time in msec after playing DTMF or MF digits to the Tel PSTN side that arrive as Relay from the IP side Valid range is 0 to 2 000 msec The default is 1 000 msec Defines the Voice Silence time in msec after detecting the end of DTMF or MF digits at the Tel PSTN side when the DTMF Transport Type is either Relay or Mute Valid range is 0 to 2 000 msec The default is 1 000 msec 356 Document LTRT 65413 SIP User s Manual Parameter Web Enable Special Digits EMS Use For Dial Termination IsSpecialDigits 6 Configuration Parameters Reference Description Determines whether the asterisk and pound digits can be used in DTMF 0 Disable Use or to terminate number collection refer to the parameter UseDigitForSpecialDTMF Default 1 Enable Allows and for telephone numbers dialed by a user or for the endpoint telephone number Note These symbols can always be used as the first digit of a dialed number even if you disable this parameter 6 16 4 RTP RTCP and T 38 Parameters The RTP RTCP and T 38 parameters are described in the table below Table 6 60 RTP RTCP and T 38 Parameters Parameter Web Dynamic Jitter Buffer Minimum Delay EMS Minimal Delay dB DJBufMinDelay Web Dynamic Jitter Buffer Optimization Factor EMS Opt Factor DJBufOptFactor Web EMS Analog Signal Transport T
411. ied by the following parameters m Burst Ring On Time Configures the cadence to be a burst cadence in the entire ringing pattern The burst relates to On time and the Off time of the same cadence It must appear between First Second Third Fourth string and the Ring On Off Time This cadence rings once during the ringing pattern Otherwise the cadence is interpreted as cyclic it repeats for every ringing cycle m Ring On Time Specifies the duration of the ringing signal m Ring Off Time Specifies the silence period of the cadence The Distinctive Ringing section of the ini file format contains the following strings m NUMBER OF DISTINCTIVE RINGING PATTERNS Contains the following key e Number of Distinctive Ringing Patterns defining the number of Distinctive Ringing signals that are defined in the file m Ringing Pattern X Contains the Xth ringing pattern definition starting from 0 and not exceeding the number of Distinctive Ringing patterns defined in the first section minus 1 using the following keys e Ring Type Must be equal to the Ringing Pattern number e Freq Hz Frequency in hertz of the ringing tone e First Burst Ring On Time 10 msec Ring On period in 10 msec units for the first cadence on off cycle e First Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the first cadence on off cycle e Second Burst Ring On Time 10 msec Ring On period in 10 msec uni
412. igital design of the filters and gain stages also ensures high reliability no drifts over temperature or time and simple variations between different line types The FXS Coefficient types provide best termination and transmission quality adaptation for two FXS line types interfaces This parameter affects the following AC and DC interface parameters DC battery feed characteristics AC impedance matching Transmit gain Receive gain Hybrid balance Frequency response in transmit and receive direction Hook thresholds Ringing generation and detection parameters FXO Operating Modes This section provides a description of the device s FXO operating modes m For IP to Tel calls refer to FXO Operations for IP to Tel Calls on page 386 m For Tel to IP calls refer to FXO Operations for Tel to IP Calls on page 390 m Call termination on FXO devices refer to Call Termination on FXO Devices on page 392 FXO Operations for IP to Tel Calls The FXO device provides the following operating modes for IP to Tel calls m One stage dialing refer to One Stage Dialing on page 387 e Waiting for dial tone refer to Two Stage Dialing on page 388 SIP User s Manual 386 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities e Time to wait before dialing e Answer supervision m Two stage dialing refer to Two Stage Dialing on page 388 m Dialing time DID wink refer to DID Wink on page 389 9 4 2 1 1 One Stag
413. ile to the device using either the BootP TFTP utility or the Web interface refer to Backing Up and Restoring Configuration on page 171 Tip Before loading the ini file to the device verify that the file extension of the ini file is correct i e ini SIP User s Manual 188 Document LTRT 65413 SIP User s Manual 4 INI File Configuration 4 3 Secured Encoded ini File The ini file contains sensitive information that is required for the functioning of the device Typically it is loaded to or retrieved from the device using TFTP or HTTP These protocols are not secure and are vulnerable to potential hackers To overcome this security threat the AudioCodes TrunkPack Downloadable Conversion Utility DConvert utility allows you to binary encode the ini file before loading it to the device refer to the Product Reference Manual If you download an ini file from the device to a folder on your PC using the Web interface refer to Backing Up and Restoring Configuration that was initially loaded to the device as encoded the file is saved encoded and vice versa Note The procedure for loading an encoded ini file is identical to the procedure for loading an unencoded ini file Version 6 0 189 March 2010 A c tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 190 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS 5 5 1 Version 6 0 Element Management System EMS
414. ination source prefixes Restricted restricts Caller ID information for these prefixes Notes Only applicable to Number Manipulation tables for Tel to IP source number manipulation If Presentation is set to Restricted and Asserted Identity Mode is set to P Asserted the From header in the INVITE message includes the following From anonymous lt sip anonymous anonymous invalid gt and privacy id header 119 March 2010 7a L tal AudioCodes MediaPack Series 3 3 4 7 3 Configuring Redirect Number Tel to IP The Redirect Number Tel gt IP page allow you to configure Tel to IP Redirect Number manipulation rules This feature manipulates the prefix of the redirect number received from the PSTN for the outgoing SIP Diversion Resource Priority or History Info header that is sent to IP You can also configure the Redirect Number Tel to IP table using the ini file parameter RedirectNumberMapTel2Ip refer to Number Manipulation and Routing Parameters on page 331 If the characteristics DestinationPrefix RedirectPrefix and or SourceAddress match the incoming SIP message manipulation is performed according to the configured manipulation rule The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and then Suffix2Add The DestinationNumber and RedirectPrefix parameters are used before any manipulation has been performed on them Redi
415. inder to later save burn your settings to flash memory and reset the device Figure 3 3 Reset Displayed on Toolbar of Submit Bun Reset Device Actions vw t Home O Help P Log off kl Reset Notification SIP User s Manual 26 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 5 Navigation Tree The Navigation tree located in the Navigation pane displays the menus pertaining to the menu tab selected on the Navigation bar used for accessing the configuration pages The Navigation tree displays a tree like structure of menus You can easily drill down to the required page item level to open its corresponding page in the Work pane The terminology used throughout this manual for referring to the hierarchical structure of the tree is as follows m menu first level highest level m submenu second level contained within a menu mpage item last level lowest level in a menu contained within a menu or submenu Figure 3 4 Terminology for Navigation Tree Levels Management K Dlagnosice Scenarios Search O Basic Full dnetwork Settings Amedia Settings security Settings Protocol Configuration Protocol Definition SIP General Parameters Proxy amp Registration Proxy Sets Table _iCoders DTMF amp Dialing sIP Advanced Parameters t manipulation Tables routing Tables Profile Definitions Wendpoint Settings t Endpoint Number BHunt 1P Group Bd
416. ing and the media streams STUN works with many existing NAT types and does not require any special behavior STUN enables the device to discover the presence and types of NATs and firewalls located between it and the public Internet It provides the device with the capability to determine the public IP address and port allocated to it by the NAT This information is later embedded in outgoing SIP SDP messages and enables remote SIP user agents to reach the device It also discovers the binding lifetime of the NAT the refresh rate necessary to keep NAT Pinholes open On startup the device sends a STUN Binding Request The information received in the STUN Binding Response IP address port is used for SIP signaling This information is updated every user defined period NATBindingDefaultTimeout At the beginning of each call and if STUN is required i e not an internal NAT call the media ports of the call are mapped The call is delayed until the STUN Binding Response that includes a global IP port for each media RTP RTCP and T 38 is received SIP User s Manual 444 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities 10 2 2 To enable STUN perform the following m Enable the STUN feature by setting the ini file parameter EnableSTUN to 1 m Define the STUN server address using one of the following methods e Define the IP address of the primary and the secondary optional STUN servers using the
417. ings to the device refer to Saving Configuration on page 161 Note This icon is grayed out when not applicable to the currently opened page Burn Saves parameter settings to flash memory refer to Saving Configuration on page 161 Device Actions w Device Opens a drop down menu list with freguently needed commands Actions Load Configuration File opens the Configuration File page for loading an ini file refer to Backing Up and Restoring Configuration on page 171 Save Configuration File opens the Configuration File page for saving the ini file to a PC refer to Backing Up and Restoring Configuration on page 171 Reset opens the Maintenance Actions page for resetting the device refer to Resetting the Device on page 159 Software Upgrade Wizard opens the Software Upgrade Wizard page for upgrading the device s software refer to Software Upgrade Wizard on page 168 Home Opens the Home page refer to Using the Home Page on page 47 G Help Opens the Online Help topic of the currently opened configuration page in the Work pane refer to Getting Help on page 45 L Log off Logs off a session with the Web interface refer to Logging Off the Web Interface on page 45 If you modify parameters that take effect only after a device reset after you click the Submit button the toolbar displays the word Reset in red color as shown in the figure below This is a rem
418. inimum Delay 40 Dynamic Jitter Buffer Optimization Factor 13 If the device initiates a fax session using G 711 option 2 and possibly 3 a gpmd attribute is added to the SDP in the following format v For A law a gpmd 8 vbd yes ecan on v For u law a gpmd 0 vbd yes ecan on When this parameter is set to 1 2 or 3 the parameter FaxTransportMode is ignored When this parameter is set to 0 T 38 might still be used without the control protocol s involvement To completely disable T 38 set FaxTransportMode to a value other than 1 For detailed information on fax transport methods refer to Fax Modem Transport Modes on page 400 ARASA Determines the default transport layer for outgoing SIP calls initiated by the device 0 UDP default 1 TCP 2 TLS SIPS Notes It s recommended to use TLS for communication with a SIP Proxy and not for direct device to device communication For received calls i e incoming the device accepts all these protocols The value of this parameter is also used by the SAS application as the default transport layer for outgoing SIP calls Local UDP port for SIP messages The valid range is 1 to 65534 The default value is 5060 Local TCP port for SIP messages The valid range is 1 to 65535 The default value is 5060 Local TLS port for SIP messages The valid range is 1 to 65535 The default value is 5061 Note The value of this parameter must be differen
419. interfaces with 10 0 0 1 8 and 10 50 10 1 24 is invalid Each interface must have its own address space The Prefix Length replaces the dotted decimal Subnet Mask presentation This column must have a value of 0 31 for IPv4 interfaces Only one IPv4 interface with OAMP Application Types must be configured At least one IPv4 interface with CONTROL Application Types must be configured At least one IPv4 interface with MEDIA Application Types must be configured These application types may be mixed i e OAMP and CONTROL Here are some examples for interface configuration e One IPv4 interface with Application Types OAMP MEDIA 8 CONTROL without VLANs e One IPv4 interface with Application Types OAMP MEDIA amp CONTROL e One IPv4 interface with Application Types OAMP one other or more IPv4 interfaces with Application Types CONTROL and one or more IPv4 interfaces with Application Types MEDIA with VLANs e One IPv4 interface with Application Types OAMP 8 MEDIA one other or more IPv4 interfaces with Application Types MEDIA amp CONTROL e Other configurations are also possible while keeping to the above mentioned rule Only one interface may have a Gateway definition for each address family IPv4 This Gateway address must be in the same subnet as this interface other routing rules may 455 March 2010 7a tal AudioCodes MediaPack Series be specified in the Routing Table Refer to Routing Table
420. ion Rate Payload Silence Time msec kbps Type Suppression G 711 A law 10 20 default Always Always 8 Disable 0 g711Alaw64k 30 40 50 60 80 64 Enable 1 100 120 G 711 U law 10 20 default Always Always 0 Disable 0 g711Ulaw64k 30 40 50 60 80 64 Enable 1 100 120 G 711A 10 20 default Always Dynamic N A law VBD 30 40 50 60 80 64 0 127 g711AlawVbd 100 120 G 711U 10 20 default Always Dynamic N A law VBD 30 40 50 60 80 64 0 127 g711UlawVbd 100 120 EG 711 A law 10 default 20 30 Always Dynamic N A eg711Alaw 64 96 127 EG 711 U law 10 default 20 30 Always Dynamic N A eg711Ulaw 64 96 127 G 722 20 default 40 64 Always 9 N A 9722 60 80 100 120 default G 723 1 30 default 60 90 5 3 0 Always 4 Disable 0 97231 6 3 1 Enable 1 default 285 March 2010 Parameter ca AudioCodes MediaPack Series Description G 726 10 20 default 16 0 Dynamic Disable 0 9726 30 40 50 60 80 default 0 127 Enable 1 100 120 24 1 Default is 32 2 23 40 3 G 727 ADPCM 10 20 default 16 24 Dynamic Disable 0 30 40 50 60 80 32 40 0 127 Enable 1 100 120 G 729 10 20 default Always Always 18 Disable 0 g729 30 40 50 60 80 8 Enable 1 100 Enable w o Adaptations 2 T 38 N A N A N A N A t38fax Notes The coder name is case sensitive Each coder type can appear only once per Coder Group Only the packetiza
421. ion when a call is made using these destination source prefixes Restricted restricts Caller ID information for these prefixes Note If Presentation is set to Restricted and Asserted Identity Mode is set to P Asserted the From header in the INVITE message includes the following From anonymous lt sip anonymous anonymous invalid gt and privacy id header 121 March 2010 7a tal AudioCodes MediaPack Series 3 3 4 7 4 Mapping NPI TON to SIP Phone Context The Phone Context Table page is used to map Numbering Plan Indication NPI and Type of Number TON to the SIP Phone Context parameter When a call is received from the Tel the NPI and TON are compared against the table and the matching Phone Context value is used in the outgoing SIP INVITE message The same mapping occurs when an INVITE with a Phone Context attribute is received The Phone Context parameter appears in the standard SIP headers where a phone number is used Request URI To From Diversion For example for a Tel to IP call with NPI TON set as E164 National values 1 2 the device sends the outgoing SIP INVITE URI with the following settings sip 12365432 phone context na e 164 nt com This is configured for entry 3 in the figure below In the opposite direction IP to Tel call if the incoming INVITE contains this Phone Context e g phone context na e 164 nt com the NPI TON of the called number in the outgoing SETUP message i
422. irewall Parameters EMS Firewall Settings AccessList SIP User s Manual This ini file table parameter configures the device s access list firewall which defines network traffic filtering rules For each packet received on the network interface the table is scanned from the top down until a matching rule is found This rule can either deny block or permit allow the packet Once a rule in the table is located subsequent rules further down the table are ignored If the end of the table is reached without a match the packet is accepted The format of this parameter is as follows ACCESSLIST FORMAT AccessList Index AccessList Source IP AccessList PrefixLen AccessList Start Port AccessList End Port AccessList Protocol AccessList Packet Size AccessList Byte Rate AccessList Byte Burst AccessList Allow Type ACCESSLIST For example AccessList 10 mgmt customer com 32 0 80 tcp 0 0 0 allow AccessList 22 10 4 0 0 16 4000 9000 any 0 0 0 block In the example above Rule 10 allows traffic from the host mgmt customer com destined to TCP ports 0 to 80 Rule 22 blocks traffic from the subnet 10 4 xxx yyy destined to ports 4000 to 9000 Notes This parameter can include up to 50 indices 232 Document LTRT 65413 SIP User s Manual Parameter 6 Configuration Parameters Reference Description To configure the firewall using the Web interface and for a description of the param
423. is considered an off hook or on hook event v Hook flash generation period upon detection of a SIP INFO message containing a hook flash signal FXO interfaces Hook flash generation period The valid range is 25 to 3 000 The default value is 700 Notes For this parameter to take effect you need to reset the device For FXO interfaces a constant of 100 msec must be added to the required hook flash period For example to generate a 450 msec hook flash set this parameter to 550 Defines when the detection of DTMF events is notified 0 DTMF event is reported at the end of a detected DTMF digit 1 DTMF event is reported at the start of a detected DTMF digit default Defines the supported Receive DTMF negotiation method 0 No Don t declare RFC 2833 telephony event parameter in SDP 3 Yes Declare RFC 2833 telephony event parameter in SDP default The device is designed to always be receptive to RFC 2833 DTMF relay packets Therefore it is always correct to include the telephony event parameter as default in the SDP However some devices use the absence of the telephony event in the SDP to decide to send DTMF digits in band using G 711 coder If this is the case you can set this parameter to 0 Determines a single or several preferred transmit DTMF negotiation methods 0 Not Supported No negotiation DTMF digits are sent according to the parameters DTMFTransportType and RF
424. it and must only be serviced by qualified service personnel Notes The following naming conventions are used throughout this manual unless otherwise specified The term device refers to the MediaPack series gateways The term MediaPack refers to the MP 124 MP 118 MP 114 and MP 112 VoIP devices The term MP 11x refers to the MP 118 MP 114 and MP 112 VoIP devices Before configuring the device ensure that it is installed correctly as instructed in the device s Installation Manual For assigning an IP address to the device refer to the device s Installation Manual FXO Foreign Exchange Office is the interface replacing the analog telephone and connects to a Public Switched Telephone Network PSTN line from the Central Office CO or to a Private Branch Exchange PBX The FXO is designed to receive line voltage and ringing current supplied from the CO or the PBX just like an analog telephone An FXO VolP device interfaces between the CO PBX line and the Internet FXS Foreign Exchange Station is the interface replacing the Exchange i e the CO or the PBX and connects to analog telephones dial up modems and fax machines The FXS is designed to supply line voltage and ringing current to these telephone devices An FXS VoIP device interfaces between the analog telephone devices and the Internet SIP User s Manual 16 Document LTRT 65413 SIP User s Manual 1 Overview 1 Overview This manual provides you
425. it for dial tone 1 Yes Wait for dial tone default When one stage dialing and this parameter are enabled the device dials the phone number to the PSTN PBX line only after it detects a dial tone If this parameter is disabled the device immediately dials the phone number after seizing the PSTN PBX line without listening for a dial tone Notes The correct dial tone parameters must be configured in the CPT 328 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web Time to Wait before Dialing msec EMS Time Before Dial WaitForDialTime Web Ring Detection Timeout sec EMS Timeout Between Rings FXOBetweenRingTime Web Rings before Detecting Caller ID EMS Rings Before Caller ID RingsBeforeCallerlD Version 6 0 Description file The device may take 1 to 3 seconds to detect a dial tone according to the dial tone configuration in the CPT file If the dial tone is not detected within 6 seconds the device releases the call and sends a SIP 500 Server Internal Error response This parameter is applicable only to FXO interfaces Determines the delay before the device starts dialing on the FXO line in the following scenarios The delay between the time the line is seized and dialing begins during the establishment of an IP to Tel call Note Applicable only for one stage dialing when the parameter IsWaitForDialTone is disabled The
426. iting indication signal When hook flash is detected by the device the device switches to the waiting call The device that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received e Disable No call waiting for the specific port e Empty Call waiting is determined according to the global parameter Enable Call Waiting described in Configuring Supplementary Services on page 111 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 142 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 10 Configuring Endpoint Phone Numbers The Endpoint Phone Number Table page allows you to activate the device s ports endpoints by defining telephone numbers for the endpoints and assigning them to Hunt Groups and profiles Each endpoint i e channel must be assigned a unique phone number In other words no two endpoints can have the same phone number You can also configure the endpoint phone numbers using the ini file table parameter TrunkGroup refer to Number Manipulation and Routing Parameters on page 331 gt To configure the Endpoint Phone Number table 1 Open the Endpoint Phone Number Table page Configuration tab gt Protocol Configuration menu gt Endpoint Number submenu gt Endpoint Phone Number page item Figure 3 90
427. ition on page 87 Proxies Registration IP Groups refer to Proxies Registrations IP Groups on page 90 Coders And Profile Definitions refer to Coders and Profile Definitions on page 101 SIP Advanced Parameters refer to SIP Advanced Parameters on page 109 Manipulation Tables refer to Manipulation Tables on page 115 Routing Tables refer to Routing Tables on page 123 Endpoint Settings refer to Endpoint Settings on page 136 Endpoint Number refer to Configuring Endpoint Phone Numbers on page 143 SAS refer to SAS Parameters on page 144 3 3 4 1 Enabling Applications The Applications Enabling page allows you to enable the Stand Alone Survivability SAS application This page displays the application only if the device is installed with the relevant Software Upgrade Key supporting the application refer to Loading a Software Upgrade Key on page 165 For enabling an application a device reset is required gt To enable an application 1 Open the Applications Enabling page Configuration tab gt Protocol Configuration menu gt Applications Enabling page item Figure 3 56 Applications Enabling Page Yv lg Enable SAS Enable 2 Save the changes to the device s flash memory and then reset the device refer to Saving Configuration on page 161 SIP User s Manual 84 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 2 Hunt
428. ive Routing table the device immediately initiates a call to the alternative destination using the next matched entry in this routing table Note that if a domain name in this table is resolved into two IP addresses the timeout for INVITE retransmissions can be reduced by using the parameter Number of RTX Before Hotswap 127 March 2010 7a e AudioCodes MediaPack Series If the alternative routing destination is the device itself the call can be configured to be routed to the PSTN This feature is referred to as PSTN Fallback For example if poor voice quality occurs over the IP network the call is rerouted through the legacy telephony system PSTN Outbound IP routing can be performed before or after number manipulation rules are applied This is configured using the RouteModeTel2IP parameter as described below You can also configure this table using the ini file table parameter Prefix refer to Number Manipulation and Routing Parameters on page 331 gt To configure Tel to IP routing rules 1 Open the Tel to IP Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Tel to IP Routing page item Figure 3 79 Tel to IP Routing Page v Routing Index 1 10 Tel To IP Routing Mode Route calls before manipulation W Src Trunk 2 Dest Group ID Dest Phone Prefix Source Phone Prefix Dest IP Address Transport Type eee IP Profile ID Status C
429. ixing and transcoding of the 3 Way Call legs on the device and even allowing multi codec conference calls The device utilizes resources from idle ports to establish the conference call The number of simultaneous on board conferences can be limited using the parameter MaxInBoardConferenceCalls In addition you can designate ports that can t be used as a resource for conference calls initiated by other ports using the parameter 3WayConfNoneAllocateablePorts Ports that are not configured with this parameter and that are idle are used by the device as a resource for establishing these type of conference calls For this mode the parameter 3WayConferenceMode is set to 2 Conferencing controlled by an external AudioCodes Conference media server The Conference initiating INVITE sent by the device uses the ConferencelD concatenated with a unique identifier as the Request URI This same Request URI is set as the Refer To header value in the REFER messages that are sent to the two remote parties For this mode the parameter 3WayConferenceMode is set to 0 default Conferencing controlled by an external third party Conference media server The Conference initiating INVITE sent by the device uses only the ConferencelD as the Request URI The Conference server sets the Contact header of the 200 OK response to the actual unique identifier Conference URI to be used by the participants This Conference URI is included by the device in the Refer T
430. jected The process of installing a client certificate on your PC is beyond the scope of this document For more information refer to your Web browser or operating system documentation and or consult your security administrator The root certificate can also be loaded via ini file using the parameter HTTPSRootFileName You can enable Online Certificate Status Protocol OCSP on the device to check whether a peer s certificate has been revoked by an OCSP server For further information refer to the Product Reference Manual 3 3 3 4 3 Self Signed Certificates The device is shipped with an operational self signed server certificate The subject name for this default certificate is ACL_nnnnnnn where nnnnnnn denotes the serial number of the device However this subject name may not be appropriate for production and can be changed while still using self signed certificates gt 1 To change the subject name and regenerate the self signed certificate Before you begin ensure the following e You have a unique DNS name for the device e g dns_name corp customer com This name is used to access the device and should therefore be listed in the server certificate e No traffic is running on the device The certificate generation process is disruptive to traffic and should be executed during maintenance time Open the Certificates page refer to Server Certificate Replacement on page 73 In the Subject Name fi
431. k The default subnet mask is 0 0 0 0 Note For this parameter to take effect a device reset is required N A Use the IP Routing table instead 209 March 2010 ca AudioCodes Parameter VLAN Parameters Web EMS VLAN Mode VLANMode Web EMS Native VLAN ID VLANNativeVLANID Web EMS OAM VLAN ID VLANOamVLANID Web EMS Control VLAN ID VLANControlVLANID Web EMS Media VLAN ID VLANMediaVLANID EnableDNSasOAM SIP User s Manual MediaPack Series Description Enables the VLAN functionality 0 Disable default 1 Enable VLAN tagging IEEE 802 1Q is enabled Notes For this parameter to take effect a device reset is required VLANs are available only when booting the device from flash When booting using BootP DHCP protocols VLANs are disabled to allow easier maintenance access In this scenario multiple network interface capabilities are not available Defines the VLAN ID to which untagged incoming traffic is assigned Outgoing packets sent to this VLAN are sent only with a priority tag VLAN ID 0 When this parameter is equal to one of the VLAN IDs in the Multiple Interface table and VLANs are enabled untagged incoming traffic is considered as incoming traffic for that interface Outgoing traffic sent from this interface is sent with the priority tag tagged with VLAN ID 0 When this parameter is different from any value in the VLAN ID column in the table untagged i
432. l 416 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities The configuration for Caller ID is described below Use the parameter CallerlDType to define the Caller ID standard Note that the Caller ID standard that is used on the PBX or phone must match the standard defined in the device Select the Bellcore caller ID sub standard using the parameter BellcoreCallerlDTypeOneSubStandard Select the ETSI FSK caller ID sub standard using the parameter ETSICallerlDTypeOneSubStandard Enable or disable per port the caller ID generation for FXS and detection for FXO using the Generate Detect Caller ID to Tel table EnableCallerlD If a port isn t configured its caller ID generation detection are determined according to the global parameter EnableCallerlD EnableCallerlDTypeTwo disables enables the generation of Caller ID type 2 when the phone is off hooked used for call waiting RingsBeforeCallerlD sets the number of rings before the device starts detection of caller ID FXO only By default the device detects the caller ID signal between the first and second rings AnalogCallerlDTimimgMode determines the time period when a caller ID signal is generated FXS only By default the caller ID is generated between the first two rings PolarityReversalType some Caller ID signals use reversal polarity and or wink signals In these scenarios it is recommended to set PolarityReversalType to 1 Hard FXS
433. l IKE SA Lifetime Remote Remote Subnet Addr Prefix Length 2 Add an Index or select the Index rule you want to edit 3 Configure the rule according to the table below SIP User s Manual 80 Document LTRT 65413 SIP User s Manual 3 Web Based Management 4 Click Apply the rule is applied on the fly 5 To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 12 IP Security Associations Table Configuration Parameters Parameter Name Operational Mode IPsecSATable_IPsecMode Remote Endopint IPsecSATable_RemoteEndpointAdd ressOrName Authentication Method IPsecSATable_AuthenticationMetho d Shared Key IPsecSATable_SharedKey Source Port IPsecSATable_SourcePort Destination Port IPsecSATable_DestPort Version 6 0 Description Defines the IPSec mode of operation 0 Transport default 1 Tunneling Defines the IP address or DNS host name of the peer Note This parameter is applicable only if the Operational Mode is set to Transport Selects the method used for peer authentication during IKE main mode 0 Pre shared Key default 1 RSA Signature in X 509 certificate Note For RSA based authentication both peers must be provisioned with certificates signed by a common CA For more information on certificates refer to Server Certificate Replacement on page 73 Defines the pre shared key in textual format Both peers
434. l Termination by PBX on page 392 m Calls terminated before call establishment refer to Call Termination before Call Establishment on page 393 m Ring detection timeout refer to Ring Detection Timeout on page 393 9 4 2 3 1 Calls Termination by PBX The FXO device supports various methods for identifying when a call has been terminated by the PBX The PBX doesn t disconnect calls but instead signals to the device that the call has been disconnected using one of the following methods m Detection of polarity reversal current disconnect The call is immediately disconnected after polarity reversal or current disconnect is detected on the Tel side assuming the PBX CO generates this signal This is the recommended method Relevant parameters EnableReversalPolarity EnableCurrentDisconnect CurrentDisconnectDuration CurrentDisconnectDefaultThreshold and TimeToSampleAnalogLineVoltage m Detection of Reorder Busy Dial and Special Information Tone SIT tones The call is immediately disconnected after a Reorder Busy Dial or SIT tone is detected on the Tel side assuming the PBX CO generates this tone This method requires the correct tone frequencies and cadence to be defined in the Call Progress Tones file If these frequencies are not known define them in the CPT file the tone produced by the PBX CO must be recorded and its frequencies analyzed refer to Adding a Reorder Tone to the CPT File in the Reference Manual
435. l is not released default 1 Enable Call is released if dial tone is detected on the device s FXO port Notes This parameter is applicable only to FXO interfaces This option is in addition to the mechanism that disconnects a call when either busy or reorder tones are detected Defines a digit pattern to send to the Tel side after a SIP 200 OK is received from the IP side The digit pattern is a user defined DTMF sequence that is used to indicate an answer signal e g for billing The valid range is 1 to 8 characters Note This parameter is applicable to FXO Determines whether the device releases the call if RTP packets are not received within a user defined timeout 0 No 1 Yes default Notes The timeout is configured by the parameter BrokenConnectionEventTimeout This feature is applicable only if the RTP session is 314 Document LTRT 65413 SIP User s Manual Parameter Web Broken Connection Timeout EMS Broken Connection Event Timeout BrokenConnectionEventTimeout Web Disconnect Call on Silence Detection EMS Disconnect On Detection Of Silence EnableSilenceDisconnect Web Silence Detection Period sec EMS Silence Detection Time Out FarEndDisconnectSilencePeriod Web Silence Detection Method FarEndDisconnectSilenceMethod Version 6 0 6 Configuration Parameters Reference Description used without Silence Compression If Silence Compression is enabled
436. l numbers from is not supported 123100 to 123200 n m Represents multiple 2 3 4 5 6 represents a one digit number numbers Up to three starting with 2 3 4 5 or 6 digits can be used to 11 22 33 xxx represents a five digit number that denote each number starts with 11 22 or 33 111 222 xxx represents a six digit number that starts with 111 or 222 n1 m1 n2 Represents a mixed 123 130 455 766 780 790 represents numbers 123 m2 a b c n3 m3 notation of multiple to 130 455 766 and 780 to 790 ranges and single numbers Note The ranges and the single numbers must have the same number of digits For example each number range and single number in the dialing plan 123 130 455 577 780 790 consists of three digits X Represents any single digit Version 6 0 377 March 2010 7a i K e AudioCodes MediaPack Series Notation Description Example Pound sign Represents the end of 54324xx represents a 7 digit number that starts with at the end of a a number 54324 number A single Represents any represents any number i e all numbers asterisk number x n l y For a description refer 0 5 3 15 to the text appearing after this table The device also supports a notation for adding a prefix where part of the prefix is first extracted from a user defined location in the original destination or source number This notation is entered in the Prefix to Add field in the Number Mani
437. l to IP Keep Alive Time 60 Alternative Routing Tone Duration ms 0 Source Manipulation Mode FROM amp PAI after manipulation v Max Allowed Packet Loss for Alt Routing 20 Max Allowed Delay for Alt Routing msec 250 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 8 3 Configuring the Tel to IP Routing The Tel to IP Routing page provides a table for configuring up to 50 Tel to IP call routing rules The device uses these rules to route calls Tel to IP destinations when a proxy server is not used for routing This table provides two main areas for defining a routing rule m Matching Characteristics user defined characteristics of the incoming call are defined in this area If the characteristics match a table entry the rule is used to route the call One or more characteristics can be defined for the rule such as Hunt Group from where the call is received source calling destination called telephone number prefix m Destination user defined IP destination If the call matches the characteristics the device routes the call to this destination The destination can be defined as an IP address or Fully Qualified Domain Name FQDN or IP Group If defined as a specific SIP User s Manual 126 Document LTRT 65413 SIP User s Manual 3 W
438. lPVersionPreference IPProfile TranscodingMode IpProfile SBCAllowedCodersGroupID IpProfile SBCAllowedCodersMode IpProfile SBCMediaSecurityBehaviour IpProfile SBCRFC2833Behavior IpProfile SBCAlternativeDTMFMethod IpProfile SBCAssertldentity IPProfile For example IPProfile 0 Sevilia 1 1 0 10 10 46 40 O 0 0 0 2 0 O 0 O 1 1 0 0 1 1 1 1 1 1 0 0 1 4294967295 0 Notes You can configure up to nine IP Profiles i e indices 1 through 9 The following parameters are not applicable SBCExtensionCodersGroupID TranscodingMode SBCAllowedCodersGroupID SBCAllowedCodersMode SBCMediaSecurityBehaviour SBCRFC2833Behavior SBCAlternativeDTMFMethod and SBCAssertldentity The parameter AddlEInSetup is not applicable The parameter MedialPVersionPreference is not applicable The parameter ISDTMFUsed is not applicable deprecated The parameter IpPreference determines the priority of the IP Profile 1 to 20 where 20 is the highest preference If both IP and Tel Profiles apply to the same call the coders and common parameters i e parameters configurable in both IP and Tel Profiles of the preferred profile are applied to that call If the Tel and IP Profiles are identical the Tel Profile parameters take precedence To use the settings of the corresponding global parameter enter the value 1 The parameter CallLimit defines the maximum number of concurrent calls allowed f
439. le 2 From the Profile ID drop down list select an identification number for the IP Profile 3 In the Profile Name field enter an arbitrary name that allows you to easily identify the IP Profile SIP User s Manual 108 Document LTRT 65413 SIP User s Manual 3 Web Based Management From the Profile Preference drop down list select the priority of the IP Profile where 1 is the lowest priority and 20 is the highest If both IP and Tel profiles apply to the same call the coders and other common parameters noted by an asterisk of the preferred Profile are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters are applied Note If the coder lists of both IP and Tel Profiles apply to the same call only the coders common to both are used The order of the coders is determined by the preference Configure the IP Profile s parameters according to your requirements Parameters that are unique to IP Profile are described in the table below From the Coder Group drop down list select the coder group that you want to assign to the IP Profile You can select the device s default coders refer to Configuring Coders on page 102 or one of the coder groups you defined in the Coder Group Settings page refer to Configuring Coder Groups on page 104 Repeat steps 2 through 6 for the next IP Profiles optional Click the Submit button to save your chang
440. le 700 Disable Not Configured Enable _ Enable 255 Disable Disable Adaptive NLP Disable wv Coder Group Coder Group Default Coder Group SIP User s Manual Document LTRT 65413 SIP User s Manual 3 Web Based Management From the Profile ID drop down list select the Tel Profile identification number you want to configure In the Profile Name field enter an arbitrary name that enables you to easily identify the Tel Profile From the Profile Preference drop down list select the priority of the Tel Profile where 1 is the lowest priority and 20 is the highest If both IP and Tel profiles apply to the same call the coders and other common parameters noted by an asterisk in the description of the parameter TelProfile of the preferred Profile are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters are applied Note If the coder lists of both IP and Tel Profiles apply to the same call only the coders common to both are used The order of the coders is determined by the preference Configure the Profile s parameters according to your requirements For detailed information on each parameter refer to its description on the page in which it is configured as an individual parameter From the Coder Group drop down list select the Coder Group refer to
441. le CallWaitingPerPort 0 CallWaitingPerPort 1 Notes 0 call waiting disabled for Port 1 1 call waiting enabled for Port 2 This parameter is applicable only to FXS ports You can configure up 8 table entries for MP 118 and up to 24 entries for MP 124 If this parameter is not configured default call waiting is determined according to the global parameter EnableCallWaiting The device s CPT file must include a call waiting Ringback tone caller side and a call waiting tone called side FXS interfaces only The EnableHold parameter must be enabled on both the calling and the called sides For configuring this table using the Web interface refer to Configuring Call Waiting on page 142 Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Number of Call Waiting indications that are played to the called telephone that is connected to the device for Call Waiting The valid range is 1 to 100 indications The default value is 2 Note This parameter is applicable only to FXS ports 295 March 2010 ca AudioCodes Parameter Web Time Between Call Waiting Indications EMS Call Waiting Time Between Indications TimeBetweenWaitingIndications Web EMS Time Before Waiting Indications TimeBeforeWaitingIndication Web EMS Waiting Beep Duration WaitingBeepDuration EMS First Call Waiting Tone ID FirstCallWaiti
442. le Table Parameters on page 186 335 March 2010 ca AudioCodes Parameter Web EMS IP to Tel Routing Mode RouteModelP2Tel Web IP Security EMS Secure Call From IP SecureCallsFromIP Web EMS Filter Calls to IP FilterCalls2IP SIP User s Manual MediaPack Series Description Determines whether to route IP calls to the Hunt Group before or after manipulation of the destination number configured in Configuring the Number Manipulation Tables on page 115 0 Route calls before manipulation Calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation Calls are routed after the number manipulation rules are applied Determines whether the device accepts SIP calls only from configured SIP Proxies or IP addresses defined in the Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 This is useful in preventing unwanted SIP calls SIP messages and or VoIP spam 0 Disable The device accepts all SIP calls default 1 Enable The device accepts SIP calls only from IP addresses defined in the Tel to IP Routing and rejects all other calls Notes When using Proxies or Proxy Sets it is unnecessary to configure the Proxy IP addresses in the routing table The device allows SIP calls received from the Proxy IP addresses even if these addresses are not configured in the routing table This feature is supported on
443. ll Detail Records CDR are sent to the Syslog server and when they are sent 0 None CDRs are not used default 1 End Call CDR is sent to the Syslog server at the end of each call 2 Start amp End Call CDR report is sent to Syslog at the start and end of each call 3 Connect 8 End Call CDR report is sent to Syslog at connection and at the end of each call 4 Start amp Connect 8 End Call CDR report is sent to Syslog at 227 March 2010 A c tal AudioCodes MediaPack Series Parameter Description the start at connection and at the end of each call Notes The CDR Syslog message complies with RFC 3161 and is identified by Facility 17 local1 and Severity 6 Informational This mechanism is active only when Syslog is enabled i e the parameter EnableSyslog is set to 1 Web EMS Debug Level Syslog debug logging level GwDebugLevel 0 0 default Debug is disabled 1 1 Flow debugging is enabled 5 5 Flow device interface stack interface session manager and device interface expanded debugging are enabled 7 7 The Syslog debug level automatically changes between level 5 level 1 and level 0 depending on the device s CPU consumption Notes Usually set to 5 if debug traces are required Options 2 3 4 and 6 are not recommended for use Web Activity Types to The Activity Log mechanism enables the device to send log messages Report via Ac
444. lls i e RouteModelP2Tel parameter is set to 0 Similar operation of removing the prefix is also achieved by using the usual number manipulation rules If enabled the device swaps the calling and called numbers received from the Tel side for Tel to IP calls The SIP INVITE message contains the swapped numbers 0 Disabled default 1 Swap calling and called numbers Determines the SIP headers containing the source number after manipulation 0 The SIP From and P Asserted Identity headers contain the source number after manipulation default 1 Only SIP From header contains the source number after manipulation while the P Asserted Identity header contains the source number before manipulation 344 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Web Add Number Plan and Type to Determines whether the TON PLAN parameters are RPI Header included in the Remote Party ID RPID header EMS Add Ton 2 RPI 0 No AddTON2RPI 1 Yes default If the Remote Party ID header is enabled EnableRPIHeader 1 and AddTON2RPI 1 it s possible to configure the calling and called number type and number plan using the Number Manipulation tables for Tel to IP calls Web Destination Phone Number Manipulation Table for Tel to IP Calls EMS SIP Manipulations gt Destination Telcom to IPs NumberMapTel2IP This ini file table parameter manipulates t
445. log Settings Page w Analog Settings Analog Metering Type 12 kHz sinusoidal bursts Jr Min Hook Flash Detection Period msec 300 Max Hook Flash Detection Period msec 700 v Coefficients Settings I FXS Coefficient Type l FXO Coefficient Type 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 64 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 2 6 Configuring Media Security The Media Security page allows you to configure media security For a detailed description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt Toconfigure media security 1 Open the Media Security page Configuration tab gt Media Settings menu gt Media Security page item Figure 3 45 Media Security Page wv General Media Security Settings Media Security Disable Media Security Behavior Preferable Disable Authentication On Transmitted RTP Packets 0 Disable Encryption On Transmitted RTP Packets Disable Encryption On Transmitted RTCP Packets wv SRTP Setting J SRTP Offered Suites Master Key Identifier MKI Size 2 Configure the parameters as required 3 Click the Submit button to save your change
446. ls EMS EMS SIP Manipulations gt Source IP to Telcom SourceNumberMapIP2Tel This ini file table parameter manipulates the source number for IP to Tel calls The format of this parameter is as follows SourceNumberMaplp2Tel FORMAT SourceNumberMaplp2Tel Index SourceNumberMaplp2Tel DestinationPrefix SourceNumberMaplp2Tel SourcePrefix SourceNumberMaplp2Tel SourceAddress SourceNumberMaplp2Tel NumberType SourceNumberMaplp2Tel NumberPlan SourceNumberMaplp2Tel RemoveFromLeft SourceNumberMaplp2Tel RemoveFromRight SourceNumberMaplp2Tel LeaveFromRight SourceNumberMaplp2Tel Prefix2Add SourceNumberMaplp2Tel Suffix2Add SourceNumberMaplp2Tel IsPresentationRestricted SourceNumberMaplp2Tel For example SourceNumberMaplp2Tel 0 22 03 5 5 2 667 5 5 SourceNumberMaplp2Tel 1 034 01 1 1 1 1 0 2 972 10 Notes The parameters NumberType NumberPlan and IsPresentationRestricted are not applicable RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add and LeaveFromRight are applied if the called and calling numbers match the DestinationPrefix SourcePrefix and SourceAddress conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and then Suffix2Add The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all addresses between 10 8 8 10 and 10 8 8 99 Th
447. lue of the parameter TimeForDialTone accordingly The MWI tone takes precedence over the Call Forwarding Reminder tone For detailed information on MWI refer to Message Waiting Indication on page 416 Determines whether the device plays a Busy Reorder tone to the PSTN side if a Tel to IP call is rejected by a SIP error response 4xx 5xx or 6xx If a SIP error response is received the device seizes the line off hook and then plays a Busy Reorder tone to the PSTN side for the duration defined by the parameter TimeForReorderTone After playing the tone the line is released on hook 0 Disable default 1 Enable Note This parameter is applicable only to FXO interfaces Duration in seconds of the Hotline dial tone If no digits are received during this duration the device initiates a call to a user defined number refer to Configuring Automatic Dialing on page 137 The valid range is 0 to 60 The default is 16 Note This parameter is applicable to FXS and FXO interfaces The delay interval in seconds from when the device receives a SIP BYE message i e remote party terminates call until the device starts playing a reorder tone to the FXS phone The valid range is 0 to 60 The default is 0 Note This parameter is applicable only to FXS interfaces 319 March 2010 ca AudioCodes Parameter Web EMS Reorder Tone Duration sec TimeForReorderTone Web EMS Enable Comfort Tone E
448. ly for numerical IP addresses in the Tel to IP Routing Enables filtering of Tel to IP calls when a Proxy is used i e IsProxyUsed parameter is set to 1 refer to Configuring Proxy and Registration Parameters on page 96 0 Don t Filter device doesn t filter calls when using a Proxy default 1 Filter Filtering is enabled When this parameter is enabled and a Proxy is used the device first checks the Tel to IP Routing before making a call through the Proxy If the number is not allowed i e number isn t listed in the table or a call restriction routing rule of IP address 0 0 0 0 is applied the call is released Note When no Proxy is used this parameter must be disabled and filtering is according to the Tel to IP Routing 336 Document LTRT 65413 SIP User s Manual Parameter Web Add CIC AddCicAsPrefix 6 Configuration Parameters Reference Description Determines whether to add the Carrier Identification Code CIC as a prefix to the destination phone number for IP to Tel calls 0 No default 1 Yes When this parameter is enabled the cic parameter in the incoming SIP INVITE can be used for IP to Tel routing decisions It routes the call to the appropriate Hunt Group based on this parameter s value For example as a result of receiving the below INVITE the destination number after number manipulation is cic 167895550001 INVITE sip 5550001 cic 16789 172 18 202 60 506
449. m Answer tones i e CED tone 0 Disabled default 1 Enabled Note This parameter is applicable only when the fax or modem transport type is set to bypass or Transparent with Events Determines the fax bypass RTP dynamic payload type The valid range is 96 to 120 The default value is 102 Modem Bypass dynamic payload type The range is 0 127 The default value is 103 Determines the fax gain control The range is 18 to 3 corresponding to 18 dBm to 3 dBm in 1 dB steps The default is 6 dBm fax gain control Defines the fax bypass output gain control The range is 31 to 31 dB in 1 dB steps The default is 0 i e no gain Defines the modem bypass output gain control The range is 31 dB to 31 dB in 1 dB steps The default is 0 i e no gain Maximum time for sending Named Telephony Events NTEs to the IP side regardless of the time range when the TDM signal is detected The range is 1 to 200 000 000 msec i e 55 hours The default is 1 i e NTE stops only upon detection of an End event Determines the basic frame size that is used during fax modem bypass sessions 0 Determined internally default 1 5 msec not recommended 2 10 msec 3 20 msec Note When set to 5 msec 1 the maximum number of simultaneous channels supported is 120 Determines the Jitter Buffer delay in milliseconds during fax and modem bypass session The range is 0 to 150 msec T
450. m audio lt udpPort A gt RTP AVP 18 0 a ptime 10 a rtpmap 96 PCMU 8000 a gpmd 96 vbd yes oOdm nod tow i ou ot a S amp S I In the example above V 152 implementation is supported using the dynamic payload type 96 and G 711 u law as the VBD codec as well as the voice codecs G 711 p law and G 729 Instead of using VBD transport mode the V 152 implementation can use alternative relay fax transport methods e g fax relay over IP using T 38 The preferred V 152 transport method is indicated by the SDP pmft attribute Omission of this attribute in the SDP content means that VBD mode is the preferred transport mechanism for voice band data To configure T 38 mode use the CodersGroup parameter SIP User s Manual 408 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 7 Working with Supplementary Services The device supports the following supplementary services Call Hold and Retrieve refer to Call Hold and Retrieve on page 409 Call Pickup refer to Call Pickup on page 411 Consultation refer to Consultation Feature on page 411 Call Transfer refer to Call Transfer on page 412 Call Forward refer to Call Forward on page 413 Call Waiting refer to Call Waiting on page 415 Message Waiting Indication refer to Message Waiting Indication on page 416 Caller ID refer to Caller ID on page 416 Three way conferencing refer to Three Way Conferencing on pa
451. m of a text password or long hex string Keys are always persisted as long hex strings and keys are localized Privacy key Keys can be entered in the form of a text password or long hex string Keys are always persisted as long hex strings and keys are localized The group with which the SNMP v3 user is associated 0 Read Only default 1 Read Write 2 Trap Note All groups can be used to send traps 3 4 1 1 4 Configuring SNMP Trusted Managers The SNMP Trusted Managers page allows you to configure up to five SNMP Trusted Managers based on IP addresses By default the SNMP agent accepts SNMP Get and Set requests from any IP address as long as the correct community string is used in the request Security can be enhanced by using Trusted Managers which is an IP address from which the SNMP agent accepts and processes SNMP requests gt To configure the SNMP Trusted Managers 1 Access the Management Settings page as described in Configuring the Management Settings on page 152 Version 6 0 157 March 2010 7a L tal AudioCodes MediaPack Series 2 In the SNMP Trusted Managers field click the right pointing arrow ua button the SNMP Trusted Managers page appears Figure 3 99 SNMP Trusted Managers Delete Trusted Managers IP Address O SNMP Trusted Manager 1 0 0 0 0 SNMP Trusted Manager 2 0 0 0 0 SNMP Trusted Manager 3 0 0 0 0 SNMP Trusted Manager 4 0 0 0 0
452. mLeft RemoveFromRight LeaveFromRight Prefix2Add and then Suffix2Add The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all addresses between 10 8 8 10 and 10 8 8 99 The Source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 To configure manipulation of destination numbers for IP to Tel calls using the Web interface refer to Configuring the Number Manipulation Tables on page 115 346 Document LTRT 65413 SIP User s Manual Parameter 6 Configuration Parameters Reference Description Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Source Phone Number Manipulation Table for Tel to IP Calls EMS SIP Manipulations gt Source Telcom to IP SourceNumberMapTel2IP Version 6 0 This ini file table parameter manipulates the source phone number for Tel to IP calls The format of this parameter is as follows SourceNumberMapTel2Ip FORMAT SourceNumberMapTel2Ip Index SourceNumberMapTel2Ip DestinationPrefix SourceNumberMapTel2Ip SourcePrefix SourceNumberMapTel2Ip SourceAddress SourceNumberMapTel2Ip NumberType SourceNumberMapTel2Ip NumberPlan SourceNumberMapTel2Ip RemoveFromLeft SourceNumberMapTel2Ip RemoveFromRight SourceNumberMapTel2Ip
453. mber Cyclic Ascending 6 By Source Phone Number Registration Mode Registration method for the Hunt Group ao CU o 1 Per Gateway Single registration for the entire device default This mode is applicable only if a default Proxy or Registrar IP are configured and Registration is enabled i e parameter IsRegisterUsed is set to 1 In this mode the SIP URI user part in the From To and Contact headers is set to the value of the global registration parameter GWRegistrationName or username if GWRegistrationName is not configured 0 Per Endpoint Each channel in the Hunt Group registers individually The registrations are sent to the ServinglPGroupID if defined in the table otherwise to the default Proxy and if no default Proxy then to the Registrar IP 4 Don t Register No registrations are sent by endpoints pertaining to the Hunt Group For example if the device is configured globally to register all its endpoints using the parameter ChannelSelectMode you can exclude some endpoints from being registered by assigning them to a Hunt Group and configuring the Hunt Group registration mode to Don t Register 5 Per Account Registrations are sent or not to an IP Group according to the settings in the Account table refer to Configuring the Account Table on page 93 SIP User s Manual 86 Document LTRT 65413 SIP User s Manual Parameter Serving IP Group ID TrunkGroupSettings_ServingIPG roup
454. md Shell link 3 Enter the following commands dr ait lt IP address of PC to collect the debug traces sent from the device gt AddChannelIdTrace ALL WITH PCM lt port number which starts from 0 gt Start 4 Make a call to the FXO 5 To stop the DR recording at the CLI prompt type STOP Caller ID on the IP Side Caller ID is provided by the SIP From header containing the caller s name and number for example From David lt SIP 101 10 33 2 2 gt tag 35dfsgasd45dg If Caller ID is restricted received from Tel or configured in the device the From header is set to From anonymous lt anonymous anonymous invalid gt tag 35dfsgasd45dg The P Asserted or P Preferred headers are used to present the originating party s caller ID even when the caller ID is restricted These headers are used together with the Privacy header m If Caller ID is restricted e The From header is set to anonymous lt anonymous anonymous invalid gt e The Privacy id header is included e The P Asserted Identity or P Preferred Identity header shows the caller ID gif Caller ID is allowed e The From header shows the caller ID e The Privacy none header is included e The P Asserted Identity or P Preferred Identity header shows the caller ID In addition the caller ID and presentation can be displayed in the Calling Remote Party ID header The Caller Display Informatio
455. mer Upon Access 300 5003 35 wv EtherDiscover Setting EtherDiscover Operation Mode Unconfigured Device Only wv IPSec Setting Enable IP Security Dead Peer Detection Mode Disable Disabled w TLS Settings TLS version TLS Client Re Handshake Interval TLS Mutual Authentication Peer Host Name Verification Mode TLS Client Verify Server Certificate TLS Remote Subject Name SSL 2 0 3 0 and TLS 1 0 0 Disable Disable Disable 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 78 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 3 7 Configuring the IP Security Proposal Table The IP Security Proposals Table page is used to configure Internet Key Exchange IKE with up to four proposal settings Each proposal defines an encryption algorithm an authentication algorithm and a Diffie Hellman group identifier The same set of proposals apply to both Main mode and Quick mode Note You can also configure the IP Security Proposals table using the ini file table parameter IPsecProposalTable refer to Security Parameters on page 232 gt To configure IP Security Proposals 1 Open the IP Security Proposals Tabl
456. message appears with the RSA host key click Yes to continue Verify that the shell prompt appears gt Type Conf and then press Enter CONFiguration gt Type cf set and then press Enter the following prompt is displayed Enter data below Type a period on an empty line to finish The configuration session is now active and all data entered at the terminal is parsed as configuration text formatted as an ini file Type the following text at the configuration session SNMPUsers FORMAT SNMPUsers Index SNMPUsers Username SNMPUsers AuthProtocol SNMPUsers PrivProtocol SNMPUsers AuthKey SNMPUsers PrivKey SNMPUsers Group SNMPUsers 0 v3user 2 1 lt auth password gt lt priv passwords 1 SNMPUsers where lt auth password gt is the password for the for the authentication protocol e lt priv password gt is the password for the privacy protocol Possible values for AuthProtocol e O none e 1 MD5 e 2 SHA 1 Possible values for PrivProtocol e O none e 1 DES e 3 AES128 To end the PuTTY configuration session type a full stop on an empty line the device responds with the following INI File replaced To save the configuration to the non volatile memory type sar the device reboots with IPSec enabled 201 March 2010 Aa L tal AudioCodes MediaPack Series 5 8 2 Configuring EMS to Operate with a Pre configured SNMPv3 System The procedure below describes
457. meter Web EMS Tel to IP Routing Mode RouteModeTel2IP Web Src Trunk Group ID EMS Source Trunk Group ID Web Dest Phone Prefix EMS Destination Phone Prefix Web EMS Source Phone Prefix 3 Web Based Management Table 3 21 Tel to IP Routing Table Parameters Description Determines whether to route received calls to an IP destination before or after manipulation of the destination number 0 Route calls before manipulation Calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation Calls are routed after the number manipulation rules are applied Notes This parameter is not applicable if outbound proxy routing is used For number manipulation refer to Configuring the Number Manipulation Tables on page 115 The Hunt Group to which the received call belongs The range is 1 99 Note To denote any Hunt Group enter an asterisk symbol Prefix of the called telephone number The prefix can include up to 50 digits Note To denote any prefix enter an asterisk symbol The prefix can be a single digit or a range of digits For available notations refer to Dialing Plan Notation for Routing and Manipulation on page 377 Prefix of the calling telephone number The prefix can include up to 50 digits Note To denote any prefix enter an asterisk symbol The prefix can be a single digit or a range of digits For available notation
458. mote Alarm Indication RAI parameters are described in the table below Parameter EnableRAI RAIHighThreshold RAlLowThreshold RAlLoopTime Table 6 15 RAI Parameters Description Enables RAI alarm generation if the device s busy endpoints exceed a user defined threshold 0 Disable RAI Resource Available Indication service default 1 RAI service enabled and an SNMP acBoardCallResourcesAlarm Alarm Trap is sent High threshold percentage of total calls that are active busy endpoints When the percentage of the device s busy endpoints exceeds this high threshold the device sends the SNMP acBoardCallResourcesAlarm alarm trap with a major alarm status The range is 0 to 100 The default value is 90 Note The percentage of busy endpoints is calculated by dividing the number of busy endpoints by the total number of enabled endpoints Low threshold percentage of total calls that are active busy endpoints When the percentage of the device s busy endpoints falls below this low threshold the device sends an SNMP acBoardCallResourcesAlarm alarm trap with a cleared alarm status The range is 0 to 100 The default value is 90 Time interval in seconds that the device periodically checks call resource availability The valid range is 1 to 200 The default is 10 6 3 4 Serial Parameters The RS 232 serial parameters are described in the table below Serial interface is mainly used for
459. mouseanonymous invalid gt tag 1c25298 To lt sip 101010 33 2 53 user phone gt tag 1c19282 CaM TDA 1102300 3342 53 CSeq 1 INVITE Contact lt sip 101 10 33 2 53 gt X Detect Response CPT FAX INFO sip 101 10 33 2 53 user phone SIP 2 0 Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 Max Forwards 70 From anonymous lt sip anonymous anonymous invalids tag 1c25298 To lt sip 101 10 33 2 53 user phone gt Call ID 11923010 334253 CSeq 1 INVITE Contact lt sip 100 10 33 2 53 gt X Detect Response CPT FAX Content Type Application X Detect Content Length xxx Type CPT Subtype SIT Version 6 0 435 March 2010 ca AudioCodes 9 12 Supported RADIUS Attributes MediaPack Series The following table provides explanations on the RADIUS attributes included in the communication packets transmitted between the device and a RADIUS Server Attribute Attribute Number Name Reguest Attributes 1 User Name 4 NAS IP Address 6 Service Type H323 26 Incoming Conf Id H323 26 Remote Address H323 Conf 26 ID 26 H323 Setu p Time 26 H323 Call Origin 26 H323 Call Type H323 26 Connect Time SIP User s Manual Table 9 5 Supported RADIUS Attributes VSA No 23 24 25 26 27 28 Purpose Account number or calling party number or blank IP address of the requesting device Type of service requested SIP call identifier IP address of the remote gateway H
460. mple AOR sip alice example com GRUU sip aliceeexample com opague kjh29x97us97d Determines whether a Cisco gateway exists at the remote side 0 No Cisco gateway exists at the remote side default 1 A Cisco gateway exists at the remote side When a Cisco gateway exists at the remote side the device must set the value of the annexb parameter of the fmtp attribute in the SDP to no This logic is used if the parameter EnableSilenceCompression is set to 2 enable without adaptation In this case Silence Suppression is used on the channel but not declared in the SDP Note The IsCiscoSCEMode parameter is applicable only when the selected coder is G 729 252 Document LTRT 65413 SIP User s Manual Parameter Web User Agent Information EMS User Agent Display Info UserAgentDisplaylnfo Web EMS SDP Session Owner SIPSDPSessionOwner Web EMS Subject SIPSubject Web Multiple Packetization Time Format EMS Multi Ptime Format MultiPtimeFormat EMS Enable P Time EnablePtime Web EMS 3xx Behavior 3xxBehavior Web EMS Enable P Charging Vector EnablePChargingVector Version 6 0 6 Configuration Parameters Reference Description Defines the string that is used in the SIP User Agent and Server response headers When configured the string lt value for UserAgentDisplaylnfo gt software version is used for example User Agent myproduct v 6 00 010 006 If not configured the
461. n table CallerDisplaylnfo is used for the following m FXS interfaces to define the caller ID per port that is sent to IP m FXO interfaces to define the caller ID per port that is sent to IP if caller ID isn t detected on the Tel side or when EnableCallerlD 0 SIP User s Manual 418 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 7 9 m FXS and FXO interfaces to determine the presentation of the caller ID allowed or restricted m To maintain backward compatibility when the strings Private or Anonymous are set in the Caller ID Name field the caller ID is restricted and the value in the Presentation field is ignored The value of the Presentation field that is defined in the Caller Display Information table can be overridden by configuring the Presentation parameter in the Tel to IP Source Number Manipulation table Therefore this table can be used to set the presentation for specific calls according to Source Destination prefixes The caller ID can be restricted allowed per port using keypad features KeyCLIR and KeyCLIRDeact FXS only AssertedidMode defines the header that is used in the generated INVITE request to deliver the caller ID P Asserted Identity or P preferred Identity Use the parameter UseTelURIForAssertedID to determine the format of the URI in these headers sip or tel The parameter EnableRPlheader enables Remote Part
462. n continuity survivability for enterprises using hosted IP services such as IP Centrex or IP PBX in cases of failure of these entities In case of failure of the IP Centrex IP PBX servers or even WAN connection and access Internet modem the enterprise typically loses its internal telephony service at any branch between its offices and with the external environment In addition typically these failures lead to the inability to make emergency calls e g 911 in North America Despite these possible point of failures the device s SAS feature ensures that the enterprise s telephony services e g SIP IP phones or soft phones are maintained by routing calls to the PSTN i e providing PSTN fallback The maximum number of SAS registered users supported by the device is 25 The SAS feature operates in one of two modes m Normal Initially the device s SAS agent serves as a registrar and an outbound Proxy server to which every VoIP CPE e g IP phones within the enterprise s LAN registers The SAS agent at the same time sends all these registration requests to the Proxy server e g IP Centrex or IP PBX This ensures registration redundancy by the SAS agent for all telephony equipment Therefore the SAS agent functions as a stateful proxy passing all SIP requests received from the enterprise to the Proxy and vice versa In parallel the SAS agent continuously maintains a keep alive handshake with the Proxy server using SIP OPTIONS
463. n File The Dial Plan file contains a list of up to eight dial plans supporting a total of up to 8 000 user defined distinct prefixes e g area codes international telephone number patterns for the PSTN to which the device is connected The Dial Plan is used for the following m Tel to IP calls The file includes up to eight patterns i e eight dial plans These allow the device to know when digit collection ends after which it starts sending all the collected or dialed digits in the INVITE message This also provides enhanced digit mapping The Dial Plan file is first created using a text based editor such as Notepad and saved with the file extension ini This ini file is then converted to a binary file dat using the DConvert utility refer to the Product Reference Manual Once converted it can then be loaded to the device using the Web interface refer to Loading Auxiliary Files on page 163 The Dial Plan file must be prepared in a textual ini file with the following syntax m Every line in the file defines a known dialing prefix and the number of digits expected to follow that prefix The prefix must be separated from the number of additional digits by a comma Empty lines are ignored m Lines beginning with a semicolon are ignored Multiple dial plans may be specified in one file a name in square brackets on a separate line indicates the beginning of a new dial plan Up to eight dial plans can be d
464. n exactly the same order m Arow ina table is identified by its table name and Index field Each such row may appear only once in the ini file m Table dependencies Certain tables may depend on other tables For example one table may include a field that specifies an entry in another table This method is used to specify additional attributes of an entity or to specify that a given entity is part of a larger entity The tables must appear in the order of their dependency i e if Table X is referred to by Table Y Table X must appear in the ini file before Table Y For general ini file formatting rules refer to General ini File Formatting Rules on page 188 The table below displays an example of an ini file table parameter CodersGroup0 FORMAT CodersGroup0 Index CodersGroup0 Name CodersGroup0 pTime CodersGroup0 rate CodersGroup0 PayloadType CodersGroup0 Sce CodersGroup0O 0 g711Alaw64k 20 ASE Ox CodersGroupO 1 eg711Ulaw 10 CodersGroupO 2 eg711Ulaw 10 CodersGroupo Note Do not include read only parameters in the ini file table parameter as this can cause an error when attempting to load the file to the device Version 6 0 187 March 2010 7a e AudioCodes MediaPack Series 4 1 3 General ini File Formatting Rules The ini file must adhere to the following format rules The ini file name must not include hyphens or spaces if necessary use an underscore _ instead Lines begin
465. n five minutes the Web session expires and you are once again requested to login with your user name and password Up to five Web users can simultaneously open log in to a session on the device s Web interface Each Web user account is composed of three attributes m User name and password enables access login to the Web interface m Access level determines the extent of the access i e availability of pages and read write privileges The available access levels and their corresponding privileges are listed in the table below Table 3 7 Web User Accounts Access Levels and Privileges Numeric Representation UES IES 200 Read write privileges for all pages read write privileges for all pages except Administiator 10 security related pages which are read only No access to security related and file loading User Monitor 50 pages read only access to the other pages This read only access level is typically applied to the secondary Web user account SIP User s Manual 66 Document LTRT 65413 SIP User s Manual 3 Web Based Management Numeric Access Level i Representation Privileges No Access 0 No access to any page The numeric representation of the access level is used only to define accounts in a RADIUS server the access level ranges from 1 to 255 The default attributes for the two Web user accounts are shown in the following table Table 3 8 Default Attributes for the Web User Accounts
466. n list select the type of device e From the Select Protocol drop down list select the the control protocol i e SIP 4 Click OK 5 In the MG Tree select the device that you want to upgrade 6 On the Actions bar click the Software Upgrade Bl button the Files Manager screen appears Figure 5 13 Files Manager Screen Files Manager Auxiliary File bav_key pem X509 PRIVAT Auxiliary File bav_signed pem x509 CERTIFI Auxiliary File new ca pem X509 TRUSTE OK J Cancel 7 Select the file that you want to download to the device and then click OK a confirmation box appears 8 Click Yes to confirm download the Software Download screen appears displaying the download progress 9 Click Done when download is completed successfully SIP User s Manual 206 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 1 Configuration Parameters Reference The device s configuration parameters default values and their descriptions are documented in this section Parameters and values enclosed in square brackets represent the ini file parameters and their enumeration values parameters not enclosed in square brackets represent their corresponding Web interface and or EMS parameters Note Some parameters are configurable only through the ini file Networking Parameters This subsection describes the device s networking parameters 6 1 1 Ethernet Parame
467. n page 143 Fora description of ini file table parameters refer to Configuring ini File Table Parameters on page 186 Version 6 0 331 March 2010 ca AudioCodes Parameter Web Hunt Group Settings EMS SIP Routing gt Hunt Group TrunkGroupSettings Web Channel Select Mode EMS Channel Selection Mode ChannelSelectMode SIP User s Manual MediaPack Series Description This ini file table parameter defines rules for channel allocation per Hunt Group If no rule exists the rule defined by the global parameter ChannelSelectMode takes effect The format of this parameter is as follows TrunkGroupSettings FORMAT TrunkGroupSettings_ Index TrunkGroupSettings_TrunkGroupld TrunkGroupSettings_ChannelSelectMode TrunkGroupSettings_RegistrationMode TrunkGroupSettings_GatewayName TrunkGroupSettings Cont actUser TrunkGroupSettings ServinglPGroup TrunkGroupSettings MWIlnterrogationType TrunkGroupSettings For example TrunkGroupSettings TrunkGroupSettings 0 1 0 5 branch hq user 1 255 TrunkGroupSettings 1 2 1 0 localname user1 2 255 TrunkGroupSettings Notes This parameter can include up to 24 indices The parameter MWIInterrogationType is not applicable For configuring Hunt Group Settings using the Web interface refer to Configuring Hunt Group Settings on page 85 Fora description on using ini file table parameters refer to to Configuring ini File Table Parameters
468. n the Vendor Specific Attributes VSA section of the received RADIUS packet The valid range is 0 to 255 The default value is 35 241 March 2010 A c tal AudioCodes MediaPack Series 6 6 SNMP Parameters The SNMP parameters are described in the table below Table 6 26 SNMP Parameters Parameter Description Web Enable SNMP Determines whether SNMP is enabled DisableSNMP 0 Enable SNMP is enabled default 1 Disable SNMP is disabled and no traps are sent SNMPPort The device s local UDP port used for SNMP Get Set commands The range is 100 to 3999 The default port is 161 Note For this parameter to take effect a device reset is required SNMPTrustedMGR_x Defines up to five IP addresses of remote trusted SNMP managers from which the SNMP agent accepts and processes SNMP Get and Set requests Notes By default the SNMP agent accepts SNMP Get and Set requests from any IP address as long as the correct community string is used in the request Security can be enhanced by using Trusted Managers which is an IP address from which the SNMP agent accepts and processes SNMP requests f no values are assigned to these parameters any manager can access the device Trusted managers can work with all community strings EMS Keep Alive Trap Port The port to which the keep alive traps are sent KeepAliveTrapPort The valid range is 0 65534 The default is port 162 SendKeepAliveTrap When enabl
469. nable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 23 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 32 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events Note This parameter applies only to V 32 and V 32bis modems V 90 V 34 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events Determines the Bell modem transport method 0 Transparent default 2 Bypass 3 Transparent with events 355 March 2010 ca AudioCodes MediaPack Series 6 16 3 DTMF Parameters The dual tone multi freguency DTMF parameters are described in the table below Parameter Web EMS DTMF Transport Type DTMFTransportType Web DTMF Volume 31 to 0 dB EMS DTMF Volume dBm DTMFVolume Web DTMF Generation Twist EMS DTMF Twist Control DTMFGenerationTwist EMS DTMF Inter Interval msec DTMFInterDigitInterval EMS DTMF Length msec DTMFDigitLength EMS Rx DTMF Relay Hang Over Time msec RxDTMFHangOverTime EMS Tx DTMF Relay Hang Over Time msec TxDTMFHangOv
470. nableComfortTone WarningToneDuration Web Play Ringback Tone to Tel EMS Play Ring Back Tone To Tel PlayRBTone2Tel SIP User s Manual MediaPack Series Description The duration in seconds that the device plays a Busy or Reorder tone duration before releasing the line The valid range is 0 to 254 The default is 0 seconds Typically after playing a Reorder Busy tone for the specified duration the device starts playing an Offhook Warning tone Notes The selection of Busy or Reorder tone is performed according to the release cause received from IP Refer also to the parameter CutThrough Determines whether the device plays a Comfort Tone Tone Type 18 to the FXS FXO endpoint after a SIP INVITE is sent and before a SIP 18x response is received 0 Disable default 1 Enable Note This parameter is applicable only to FXO FXS interfaces Defines the duration in seconds for which the Off Hook Warning Tone is played to the user The valid range is 1 to 2 147 483 647 The default is 600 Note A negative value indicates that the tone is played infinitely Enables the play of the ringback tone RBT to the Tel side and determines the method for playing the RBT 0 Don t Play RBT is not played 1 Play Local RBT is played to the Tel side of the call when a SIP 180 183 response is received 2 Play According to Early Media RBT is played to the Tel side only if a 180 183 response
471. name and password the EMS server s IP address and then click OK 3 Add a Region for your deployed device by performing the following SIP User s Manual 194 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS a Inthe MG Tree right click the Globe 3 icon and then click Add Region the Region dialog box appears Figure 5 3 Adding a Region Region Name Description Lx JL cancel b In the Region Name field enter a name for the Region e g a geographical name and then click OK the Region is added to the MG Tree list 4 Verify that the device is up and running by performing a ping to its IP address 5 Add the device to the Region by performing the following a Right click the added Region Hf icon and then from the shortcut menu choose Add MG the MG Information dialog box appears Figure 5 4 Defining the IP Address MG Information General 3 SNMPv2 SNMPv3 SNMP MG Name SNMP Read Community public IP Address SNMP Write Communi ivat Description ty private OAM Secure Connection IPSec Enabled IKE Pre Shared Key OK Cancel x b Enter an arbitrary name for the device and then in the IP Address field enter the device s IP address c Ensure that IPSec Enabled check box is selected and then enter the IPSec Preshared Key defined in Configuring IPSec on page 192 d Click OK the device is added to the Region
472. nce Statistics Page Statistics for 2811 seconds Active TDM channels Active DSP resources Active analog channels Active G 711 channels Average voice delay ms Average voice jitter ms Total RTP packets TX Total RTP packets RX Total call attempts Reset Statistics gt To reset the performance statistics to zero m Click the Reset Statistics button SIP User s Manual 176 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 5 1 6 Viewing Active Alarms The Active Alarms page displays a list of currently active alarms You can also access this page from the Home page refer to Using the Home Page on page 47 gt To view the list of alarms m Open the Active Alarms page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Active Alarms page item Figure 3 113 Active Alarms Page For each alarm the following information is provided m Severity severity level of the alarm e Critical alarm displayed in red e Major alarm displayed in orange e Minor alarm displayed in yellow m Source unit from which the alarm was raised m Description brief explanation of the alarm m Date date and time that the alarm was generated You can view the next 30 alarms if exist by pressing the F5 key Version 6 0 177 March 2010 7a K tal AudioCodes MediaPack Series 3 5 2 3 5 2 1 Gateway Statistics The Gateway Sta
473. nce over the second coder and so on The first coder is the highest priority coder and is used by the device whenever possible If the far end device cannot use the coder assigned as the first coder the device attempts to use the next coder and so on For a list of supported coders and for configuring coders using the ini file refer to the ini file parameter table CodersGroup described in SIP Configuration Parameters on page 245 Each coder type can appear only once per Coder Group The device always uses the packetization time reguested by the remote side for sending RTP packets If not specified the packetization time ptime is assigned the default value Only the packetization time of the first coder in the defined coder list is declared in INVITE 200 OK SDP even if multiple coders are defined For G 729 you can also select silence suppression without adaptations If silence suppression is enabled for G 729 the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is set to Enable w o Adaptations annexb yes is included An exception is when the remote device is a Cisco gateway IsCiscoSCEMode gt To configure coder groups 1 Open the Coder Group Settings page Configuration tab gt Protocol Configuration menu gt Coders And Profile Definitions submenu gt Coder Group Settings page item Figure 3 65 Coder Group Settings Page vw Coder Group ID
474. ncoming traffic is discarded and all outgoing traffic is tagged Note If this parameter is not set i e default value is 1 but one of the interfaces has a VLAN ID configured to 1 this interface is still considered the Native VLAN If you do not wish to have a Native VLAN ID and want to use VLAN ID 1 set this parameter to a value other than any VLAN ID in the table Defines the OAMP VLAN identifier The valid range is 1 to 4094 The default value is 1 Defines the Control VLAN identifier The valid range is 1 to 4094 The default value is 2 Defines the Media VLAN identifier The valid range is 1 to 4094 The default value is 3 This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for DNS services VLANs Determines the traffic type for DNS services 1 OAMP default 0 Control Note For this parameter to take effect a device reset is required 210 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description EnableNTPasOAM This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for NTP services VLANs Determines the traffic type for NTP services 1 OAMP default 0 Control Note For this parameter to take effect a device reset is required VLANSendNonTaggedOnNative Determines whether to send non tagged packets on the nati
475. nd MWI default 1 Standard ETSI Caller ID and MWI 2 Standard NTT 4 Standard BT Britain 16 Standard DTMF Based ETSI 17 Standard Denmark Caller ID and MWI 18 Standard India 19 Standard Brazil Notes Typically the Caller ID signals are generated detected between the first and second rings However sometimes the Caller ID is detected before the first ring signal in such a scenario configure the parameter RingsBeforeCallerlD to 0 Caller ID detection for Britain 4 is not supported on the device s FXO ports Only FXS ports can generate the Britain 4 Caller ID To select the Bellcore Caller ID sub standard use the parameter BellcoreCallerlDTypeOneSubStandard To select the ETSI Caller ID substandard use the parameter ETSICallerlDTypeOneSubStandard To select the Bellcore MWI sub standard use the parameter Bellcore VMWITypeOneStandard To select the ETSI MWI sub standard use the parameter ETSIVMWITypeOneStandard If you define Caller ID Type as NTT 2 you need to define the NTT DID signaling form FSK or DTMF using the parameter NTTDIDSignallingForm 291 March 2010 ca AudioCodes Parameter Web Enable FXS Caller ID Category Digit For Brazil Telecom AddCPCPrefix2BrazilCallerlD EnableCallerIDTypeTwo SIP User s Manual MediaPack Series Description Enables the interworking of Calling Party Category cpc code from SIP INVITE messages to FXS Caller ID fir
476. nd for configuring the table using the Web interface refer to Configuring SNMP V3 Users on page 156 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 244 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 7 SIP Configuration Parameters This subsection describes the device s SIP parameters 6 7 1 General SIP Parameters The general SIP parameters are described in the table below Table 6 27 General SIP Parameters Parameter SIPForceRport Web Max Number of Active Calls EMS Maximum Concurrent Calls MaxActiveCalls Web EMS PRACK Mode PrackMode Web EMS Enable Early Media EnableEarlyMedia Version 6 0 Description Determines whether the device sends SIP responses to the UDP port from where SIP requests are received even if the rport parameter is not present in the SIP Via header 0 default Disabled the device sends the SIP response to the UDP port defined in the Via header If the Via header contains the rpor parameter the response is sent to the UDP port from where the SIP reguest is received 1 Enabled SIP responses are sent to the UDP port from where SIP requests are received even if the rpor parameter is not present in the Via header Defines the maximum number of simultaneous active calls supported by the device If the maximum number of calls is reached new calls are
477. nd modem signals are transferred using Cisco compatible Pass through bypass mode Upon detection of fax or modem answering tone signal the terminating device sends three to six special NSE RTP packets using NSEpayloadType usually 100 These packets signal the remote device to switch to G 711 coder according to the parameter FaxModemBypassCoderType After a few NSE packets are exchanged between the devices both devices start using G 711 packets with standard payload type 8 for G 711 A Law and 0 for G 711 Mu Law In this mode no Re INVITE messages are sent The voice channel is optimized for fax modem transmission Same as for usual bypass mode The parameters defining payload type for the proprietary AudioCodes Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass When configured for NSE mode the device includes in its SDP the following line a rtpmap 100 X NSE 8000 where 100 is the NSE payload type The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw To configure NSE mode perform the following configurations IsFaxUsed 0 FaxTransportMode 2 NSEMode 1 NSEPayloadType 100 V21ModemTransportType 2 V22ModemTransportType 2 V23ModemTransportType 2 V32ModemTransportType 2 SIP User s Manual 404 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 6 2 6 9 6 2 7 m V34ModemTransportType
478. ne Num Inter Digit Teneout for Overlap Dising sec Declare RFC 2033 in SDP ist Tx DTMF Option 2nd Tx DTMF Option 3rd Tx DTMF Option 4th Tx OTMF Option Sth Tx DTMF Option RFC 2633 Payload Type Digit Mapping Rules MediaPack Series Drouting Tables ull Profle Oefirebons ubendoown Settines Scenario Name PBX Interoperability i Define Coders Deal Tone Durston sec Hotline Dial Tone Duration sec Enedle Special Digits Default Desbnation Number Special Digit Representation Added Scenario Step Scenario Name PEX Infeeoper aday Step Namo SIPPODIME Defining Step Name Defining Scenario Name Save Firish Cancet Scenarios Geti Send Scenario Fie 8 Repeat steps 5 through 8 to add additional Steps i e pages 9 When you have added all the required Steps for your Scenario click the Save amp Finish button located at the bottom of the Navigation tree a message box appears informing you that the Scenario has been successfully created 10 Click OK the Scenario mode is quit and the menu tree of the Configuration tab appears in the Navigation tree You can add up to 20 Steps to a Scenario where each Step can contain up to 25 parameters When in Scenario mode the Navigation tree is in Full display i e all menus are displayed in the Navigation tree and the configuration pages are in Advanced Parameter List display i e all parameters are shown in the pages This ensu
479. nes the digit pattern which upon detection generates the Conference initiating INVITE when 3 way conferencing is enabled Enable3WayConference is set to 1 The valid range is a 25 character string The default is Hook Flash Defines the Conference Identification string up to 16 characters The default value is conf The device uses this identifier in the Conference initiating INVITE that is sent to the media server when Enable3WayConference is set to 1 For example ConferencelD MyConference 304 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 8 8 Emergency Call Parameters The emergency call parameters are described in the table below Table 6 41 Emergency Call Parameters Parameter Description Web EMS Emergency Defines a list of numbers which are defined as emergency Numbers numbers When one of these numbers is dialed the outgoing INVITE EmergencyNumbers message includes the Priority and Resource Priority headers If the user sets the phone on hook the call is not disconnected but instead a Hold Re INVITE request is sent to the remote party Only if the remote party disconnects the call i e a BYE is received or a timer expires set by the parameter EmergencyRegretTimeout is the call terminated The list can include up to four different numbers where each number can be up to four digits long Example EmergencyNumbers 100 911 112 Note This parameter is applic
480. nforming you of the new cmp file 12 Click OK the Web interface becomes active reflecting the upgraded device 3 4 2 4 Backing Up and Restoring Configuration You can save a copy backup of the device s current configuration settings as an ini file to a folder on your PC using the Configuration File page The saved ini file includes only parameters that were modified and parameters with other than default values The Configuration File page also allows you to load an ini file to the device If the device has lost its configuration you can restore the device s configuration by loading the previously saved ini file or by simply loading a newly created ini file Note When loading an ini file using this Web page parameters not included in the ini file are reset to default settings gt To save and restore the ini file 1 Open the Configuration File page Management tab gt Software Update menu gt Configuration File Figure 3 109 Configuration File Page Save the INI file to the PC Save INI File Send the INI file to the device CEES The device will perform a reset after sending the INI file 2 To save the ini file to a folder on your PC perform the following 3 Click the Save INI File button the File Download dialog box appears 4 Click the Save button navigate to the folder in which you want to save the ini file on your PC and then click Save the device copies the ini file to the
481. ng and restoring configuration refer to Backing Up and Restoring Configuration on page aree The Software Upgrade Wizard requires the device to be reset at the end of the process which may disrupt traffic To avoid this disable all traffic on the device before initiating the wizard by performing a graceful lock refer to Saving and Resetting the Device SIP User s Manual 168 Document LTRT 65413 SIP User s Manual 3 Web Based Management Before you can load an ini or any auxiliary file you must first load a cmp file When you activate the wizard the rest of the Web interface is unavailable After the files are successfully loaded access to the full Web interface is restored If you upgraded your cmp and the SW version mismatch message appears in the Syslog or Web interface you know that your Software Upgrade Key does not support the new cmp version Contact AudioCodes support for assistance You can schedule automatic loading of these files using HTTP HTTPS FTP or NFS refer to the Product Reference Manual gt To load files using the Software Upgrade Wizard 1 Stop all traffic on the device using the Graceful Lock feature refer to the warning bulletin above 2 Open the Software Upgrade Wizard Management tab gt Software Update menu gt Software Upgrade Wizard the Software Upgrade Wizard page appears Figure 3 107 Start Software Upgrade Wizard Screen Start Software Upgrade Click the b
482. ng rules for calls from Hunt Group 1 to IP Group 1 and from Hunt Group 2 to IP Group 2 Figure 9 31 Configuring Hunt Group to ITSP Routing Dest Src Trunk Dest IP Address Transport Type IPGroup ID Group ID Dest Phone Prefix Source Phone Prefix Not Configured vi l m i Nat Configured v 2 v Version 6 0 431 March 2010 A e AudioCodes MediaPack Series 9 9 9 10 Mapping PSTN Release Cause to SIP Response The device s FXO interface interoperates between the SIP network and the PSTN PBX This interoperability includes the mapping of PSTN PBX Call Progress Tones to SIP 4xx or 5xx responses for IP to Tel calls The converse is also true for Tel to IP calls the SIP 4xx or 5xx responses are mapped to tones played to the PSTN PBX When establishing an IP to Tel call the following rules are applied m If the remote party PSTN PBX is busy and the FXO device detects a Busy tone it sends a SIP 486 Busy response to IP If it detects a Reorder tone it sends a SIP 404 Not Found no route to destination to IP In both cases the call is released Note that if the parameter DisconnectOnBusyTone is set to 0 the FXO device ignores the detection of Busy Reorder tones and doesn t release the call m For all other FXS FXO release types caused when there are no free channels in the specific Hunt Group or when an appropriate rule for routing the call to a Hunt Group doesn t e
483. ngTonelD SIP User s Manual MediaPack Series Description Time in seconds between consecutive call waiting indications for call waiting The valid range is 1 to 100 The default value is 10 Note This parameter is applicable only to FXS ports Defines the interval in seconds before a call waiting indication is played to the port that is currently in a call The valid range is 0 to 100 The default time is 0 seconds Note This parameter is applicable only to FXS ports Duration in msec of call waiting indications that are played to the port that is receiving the call The valid range is 100 to 65535 The default value is 300 Note This parameter is applicable only to FXS ports Determines the index of the first Call Waiting Tone in the CPT file This feature enables the called party to distinguish between different call origins e g external versus internal calls There are three ways to use the distinctive call waiting tones Playing the call waiting tone according to the SIP Alert Info header in the received 180 Ringing SIP response The value of the Alert Info header is added to the value of the FirstCallWaitingTonelD parameter Playing the call waiting tone according to Prioritylndex in the Tonelndex ini file table parameter Playing the call waiting tone according to the parameter CallWaitingTone of a SIP INFO message The device plays the tone received in the play tone CallWaitingTone pa
484. ning with a semi colon are ignored These can be used for adding remarks in the ini file A carriage return i e Enter must be done at the end of each line The number of spaces before and after the equals sign is irrelevant Subsection names for grouping parameters are optional If there is a syntax error in the parameter name the value is ignored Syntax errors in the parameter s value can cause unexpected errors parameters may be set to the incorrect values Parameter string values that denote file names e g CallProgressTonesFileName must be enclosed with inverted commas e g CallProgressTonesFileName cpt_usa dat The parameter name is not case sensitive The parameter value is not case sensitive except for coder names The ini file must end with at least one carriage return 4 2 Modifying an ini File You can modify an ini file currently used by the device Modifying an ini file instead of loading an entirely new ini file preserves the device s current configuration including factory default values gt 1 To modify an ini file Save the current ini file from the device to your PC using the Web interface refer to Backing Up and Restoring Configuration on page 171 Open the ini file using a text file editor such as Microsoft Notepad and then modify the ini file parameters according to your requirements Save the modified ini file and then close the file Load the modified ini f
485. no communication is detected for about three minutes the device performs a self test If the self test succeeds the problem is a logical link down i e Ethernet cable disconnected on the switch side and the Busy Out mechanism is activated if enabled i e the parameter EnableBusyOut is set to 1 Lifeline is activated only if it is enabled using the parameter LifeLineType If the self test fails the device restarts to overcome internal fatal communication error Notes For this parameter to take effect a device reset is required Enable LAN Watchdog is relevant only if the Ethernet connection is full duplex LAN Watchdog is not applicable to MP 118 0 Disable device s watch dog 1 Enable device s watch dog default Note For this parameter to take effect a device reset is required Defines the scenario upon which the Lifeline phone is activated The Lifeline phone is available on Port 1 of MP 11x FXS devices and on ports 1 to 4 of MP 118 FXS FXO devices For FXS only devices FXS Port 1 is connected to the POTS Lifeline phone as well as to the 225 March 2010 A c tal AudioCodes MediaPack Series Parameter Web Delay After Reset sec GWAppDelayTime Description PSTN PBX using a splitter cable For combined FXS and FXO devices the FXS ports are provided with lifeline by their corresponding FXO ports connected to the PSTN PBX i e FXO Port 5 provides lifeline to FXS Port 1 FXO
486. nored For a detailed description of the digit mapping refer to Digit Mapping on page 379 Defines the maximum number of collected destination number digits that can be received i e dialed from the Tel side When the number of collected digits reaches this maximum the device uses these digits for the called destination number The valid range is 1 to 49 The default value is 5 Notes Digit Mapping Rules can be used instead Dialing ends when any of the following scenarios occur y Maximum number of digits is dialed v Interdigit Timeout TimeBetweenDigits expires v Pound key is pressed v Digit map pattern is matched Defines the time in seconds that the device waits between digits that are dialed by the user When this inter digit timeout expires the device uses the collected digits to dial the called destination number The valid range is 1 to 10 The default value is 4 6 7 7 Coders and Profile Parameters The profile parameters are described in the table below Parameter Table 6 33 Profile Parameters Description Web Coders Table Coder Group Settings EMS Coders Group CodersGroup0 CodersGroup1 CodersGroup2 CodersGroup3 CodersGroup4 SIP User s Manual This ini file table parameter defines the device s coders Up to five groups of coders can be defined where each group can consist of up to 10 coders The first Coder Group is the default coder list and the default Coder Group
487. nsions to IP numbers This file can be used to represent PBX extensions as IP phones in the global IP world You can schedule automatic loading of updated auxiliary files using HTTP HTTPS FTP or NFS refer to the Product Reference Manual For a detailed description on auxiliary files refer to Auxiliary Configuration Files on page 367 When loading an ini file the current settings of parameters that are excluded from the loaded ini file are retained incremental Saving an auxiliary file to flash memory may disrupt traffic on the device To avoid this disable all traffic on the device by performing a graceful lock refer to Locking and Unlocking the Device on page 161 For deleting auxiliary files refer to Viewing Device Information on page 174 Version 6 0 163 March 2010 7a c tal AudioCodes MediaPack Series The auxiliary files can be loaded to the device using the Web interface s Load Auxiliary Files page as described in the procedure below gt To load an auxiliary file to the device using the Web interface 1 Open the Load Auxiliary Files page Management tab gt Software Update menu gt Load Auxiliary Files page item Figure 3 104 Load Auxiliary Files Page Load auxiliary Files ooo SA an A INI file incremental Call Progress Tones file Prerecorded Tones file Dial Plan file User Info file 2 Click the Brows
488. nt subscription for Ring reminder event notification feature 0 Disable default 1 Enable Defines the IP Group ID that contains the Application server for Subscription The valid value range is 1 to 8 The default is 1 i e not configured Defines the Retry period in seconds for Dialog subscription if a previous request failed The valid value range is 10 to 7200 The default is 120 Defines the ringing tone type played when call forward notification is accepted The valid value range is 1 to 5 The default is 1 298 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 8 4 Message Waiting Indication Parameters The message waiting indication MWI parameters are described in the table below Parameter Web Enable MWI EMS MWI Enable EnableMWI Web EMS MWI Analog Lamp MWIAnalogLamp Web EMS MWI Display MWIDisplay Web Subscribe to MWI EMS Enable MWI Subscription EnableMWISubscription Web MWI Server IP Address EMS MWI Server IP MWIServerlP Version 6 0 Table 6 37 MWI Parameters Description Enables Message Waiting Indication MWI 0 Disable Disabled default 1 Enable MWI service is enabled Notes This parameter is applicable only to FXS interfaces The device supports only the receipt of SIP MWI NOTIFY messages the device doesn t generate these messages For detailed information on MWI refer to Message Waiting
489. ntry m Apply saves the configuration gt To add an entry to a table 1 In the Add Index field enter the desired index entry number and then click Add Index an index entry row appears in the table Figure 3 11 Adding an Index Entry to a Table Entered index Add Index Button Number _ Index Application Type IP Address Prefix VLAN Length Gateway ID Interface Name 0 GAMP Media Control w 1013413 he 1013 01 I Jom Added Index Row 2 Click Apply to save the index entry Before you can add another index entry you must ensure that you have applied the previously added index entry by clicking Apply If you leave the Add field blank and then click Add Index the existing index entries are all incremented by one and the newly added index entry is assigned the index 0 gt To add a copy of an existing index table entry 1 In the Index column select the index that you want to duplicate the Edit button appears 2 Click Edit the fields in the corresponding index row become available 3 Click Duplicate a new index entry is added with identical settings as the selected index in Step 1 In addition all existing index entries are incremented by one and the newly added index entry is assigned the index 0 SIP User s Manual 34 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To edit an existing index table entry 1 In the Index column select the inde
490. number to indicate an external line This number 9 can then be removed by number manipulation before the call is setup m Allowing or blocking Caller ID information to be sent according to destination or source prefixes For detailed information on Caller ID refer to Configuring Caller Display Information on page 138 The number manipulation is configured in the following tables m For Tel to IP calls e Destination Phone Number Manipulation Table for Tel to IP Calls NumberMapTelZ2IP ini file parameter up to 120 entries e Source Phone Number Manipulation Table for Tel to IP Calls SourceNumberMapTel2IP ini file parameter up to 20 entries m For IP to Tel calls e Destination Phone Number Manipulation Table for IP to Tel Calls NumberMapIP2Tel ini file parameter up to 100 entries e Source Phone Number Manipulation Table for IP to Tel Calls SourceNumberMapIP2Tel ini file parameter up to 20 entries The device matches manipulation rules starting at the top of the table In other words a rule at the top of the table takes precedence over a rule defined lower down in the table Therefore define more specific rules above more generic rules For example if you enter 551 in Index 1 and 55 in Index 2 the device applies rule 1 to numbers that start with 551 and applies rule 2 to numbers that start with 550 552 553 and so on untill 559 However if you enter 55 in Index 1 and 551 in Index 2 the device applies rule 1
491. nvalid in the INVITE s From header for Tel to IP calls 0 default If the device receives a call from the Tel with blocked caller ID it sends an INVITE with From anonymous lt anonymous anonymous invalid gt 1 The device s IP address is used as the URI host part instead of anonymous invalid This parameter may be useful for example for service providers who identify their SIP Trunking customers by their source phone number or IP address reflected in the From header of the SIP INVITE Therefore even customers blocking their Caller ID can be identified by the service provider Typically if the device receives a call with blocked Caller ID from the PSTN side e g Trunk connected to a PBX it sends an INVITE to the IP with a From header as follows From anonymous lt anonymous anonymous invalid gt This is in accordance with RFC 3325 However when this parameter is set to 1 the device replaces the anonymous invalid with its IP address Defines a representative number up to 50 characters that is used as the user part of the Request URI in the P Asserted Identity header of an outgoing INVITE for Tel to IP calls The default value is null Defines the source for the SIP URI set in the Refer To header of outgoing REFER messages 0 Use SIP URI from Contact header of the initial call default 1 Use SIP URI from To From header of the initial call Enables or disables the usag
492. o 123456789 In addition a lifetime of 28800 seconds is selected for IKE and a lifetime of 3600 seconds is selected for IPSec Notes Each row in the table refers to a different IP destination To support more than one Encryption Authentication proposal for each proposal specify the relevant parameters in the Format line The proposal list must be contiguous Fora detailed description of this table and to configure the table using the Web interface refer to Configuring the IP Security Associations Table on page 80 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 SIP User s Manual 238 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Web IP Security Proposal Table EMS IPSec Proposal Table IPSecProposalTable This ini file table parameter configures up to four IKE proposal settings where each proposal defines an encryption algorithm an authentication algorithm and a Diffie Hellman group identifier IPsecProposalTable FORMAT IPsecProposalTable Index IPsecProposalTable EncryptionAlgorithm IPsecProposalTable AuthenticationAlgorithm IPsecProposalTable DHGroup WPsecProposalTable For example IPsecProposalTable 0 3 2 1 IPsecProposalTable 1 2 2 1 In the example above two proposals are defined Proposal 0 AES SHA1 DH group 2 Proposal 1 3DES SHA
493. o Call Waiting on page 415 For information on the Call Progress Tones file refer to Configuring the Call Progress Tones File Determines the SIP response code for indicating Call Waiting 0 Use 182 Queued response to indicate call waiting default 1 Use 180 Ringing response to indicate call waiting 294 Document LTRT 65413 SIP User s Manual Parameter Web Call Waiting Table EMS SIP Endpoints gt Call Waiting CallWaitingPerPort Web Number of Call Waiting Indications EMS Call Waiting Number of Indications NumberOfWaitingIndications Version 6 0 6 Configuration Parameters Reference Description This ini file table parameter configures call waiting per FXS port The format of this parameter is as follows CallWaitingPerPort FORMAT CallWaitingPerPort_Index CallWaitingPerPort_IsEnabled CallWaitingPerPort Where Index port number where 0 denotes Port 1 IsEnabled v 0 Disable no call waiting for the specific port y 1 Enable enables call waiting for the specific port When the FXS device receives a call on a busy endpoint port it responds with a SIP 182 response and not with a 486 busy The device plays a call waiting indication signal When hook flash is detected the device switches to the waiting call The device that initiates the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received For examp
494. o configure the IP Groups table using the ini file table parameter IPGroup refer to SIP Configuration Parameters on page 245 gt To configure IP Groups 1 Open the IP Group Table page Configuration tab gt Protocol Configuration menu gt Proxies Registration IP Groups submenu gt IP Group Table page item Figure 3 60 IP Group Table Page 1 Common Parameters Description Proxy Set ID SIP Group Name Contact User IP Profile ID Gateway Parameters Always Use Route Table Yes Routing Mode Serving IP Group SIP Re Routing Mode Standard Enable Survivability Disable Serving IP Group ID 3 2 Configure the IP group parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 91 March 2010 ca AudioCodes MediaPack Series Table 3 14 IP Group Parameters Parameter Common Parameters Description IPGroup Description Proxy Set ID IPGroup ProxySetld SIP Group Name IPGroup_SIPGroupName Contact User IPGroup_ContactUser IP Profile ID IPGroup Profileld Gateway Parameters Always Use Route Table IPGroup AlwaysUseRouteTable SIP Re Routing Mode IPGroup_SIPReRoutingMode SIP User s Manual Description Brief string description of the IP Group The value range is a string of up to 29 characters Th
495. o header value in the REFER messages sent by the device to the remote parties The remote parties join the conference by sending INVITE messages to the Conference server using this conference URI For this mode the parameter 3WayConferenceMode is set to 1 To enable three way conferencing the following parameters need to be configured Enable3WayConference ConferenceCode default which is the hook flash button HookFlashCode 3WayConferenceMode conference mode MaxInBoardConferenceCalls if on board conferencing 3WayConfNoneAllocateablePorts if on board conferencing FlashKeysSequenceStyle 1 makes a three way call conference using the Flash button 3 SIP User s Manual 420 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 8 Routing Examples 9 8 1 SIP Call Flow Example The SIP call flow shown in the following figure describes SIP messages exchanged between two devices during a basic call In this call flow example device 10 8 201 158 with phone number 6000 dials device 10 8 201 161 with phone number 2000 Figure 9 19 SIP Call Flow 10 8 201 108 10 8 201 161 Phone 6000 Phone 2000 INVITE F1 100Trying F2 m F1 INVITE 10 8 201 108 gt gt 10 8 201 161 INVITE sip 2000 10 8 201 161 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 6000e10 8 201 108 gt tag 1c5354 To lt sip 2000 10 8 201 161 gt Call ID 5343665566
496. of a parameter click the plus sign to expand the parameter To collapse the description click the minus sign To close the Help topic click the close button located on the top right corner of the Help topic window Instead of clicking the Help button for each page you open you can open it once for a page and then simply leave it open Each time you open a different page the Help topic pertaining to that page is automatically displayed Version 6 0 45 March 2010 7a E tal AudioCodes MediaPack Series 3 1 11 Logging Off the Web Interface You can log off the Web interface and re access it with a different user account For detailed information on the Web User Accounts refer to User Accounts gt To log off the Web interface 1 On the toolbar click the Log Off lt button the Log Off confirmation message box appears Figure 3 24 Log Off Confirmation Box Microsoft Internet Explorer 2 j Logoff 2 Click OK the Web session is logged off and the Log In button appears Figure 3 25 Web Session Logged Off Zi http 10 13 4 13 HiddenPressl ogOff Microsoft Interne E JE af File Edit view Favorites Tools Help Bak Address a http 10 13 4 13 HiddenPressLogOff Web session is logged off Internet To log in again simply click the Log In button and then in the Enter Network Password dialog box enter your user name and password refer to Accessing the Web Interface
497. of all the Web user accounts Web users with an access level other than Security Administrator can only change their own password and user name To reset the two Web user accounts user names and passwords to default set the ini file parameter ResetWebPassword to 1 To access the Web interface with a different account click the Log off button located on the toolbar click any button or page item and then re access the Web interface with a different user name and password You can set the entire Web interface to read only regardless of Web user account s access level by using the ini file parameter DisableWebConfig refer to Web and Telnet Parameters on page 222 Access to the Web interface can be disabled by setting the ini file parameter DisableWebTask to 1 By default access is enabled You can define additional Web user accounts using a RADIUS server refer to the Product Reference Manual For secured HTTP connection HTTPS refer to the Product Reference Manual 68 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 3 2 Configuring the Web and Telnet Access List The Web amp Telnet Access List page is used to define up to ten IP addresses that are permitted to access the device s Web and Telnet interfaces Access from an undefined IP address is denied If no IP addresses are defined this security feature is inactive and the device can be accessed from any IP address The Web and Teln
498. oft s application with the URI user part as INVITE sip 622125519100 ext 104 10 1 1 10 or INVITE tel 622125519100 ext 104 If the parameter EnableMicrosofExt is enabled the device modifies the called number by adding an e as the prefix removing the ext parameter and adding the extension number as the suffix e g e622125519100104 Once modified the device can then manipulate the number further using the Number Manipulation tables refer to Number Manipulation and Routing Parameters on page 331 to leave only the last 3 digits for example for sending to a PBX Defines the URI format in the SIP Diversion header 0 tel default 1 sip Defines the timeout in msec between receiving a 100 Trying response and a subsequent 18x response If a 18x response is not received before this timer expires the call is disconnected The valid range is 0 to 32 000 The default value is 0 i e no timeout Enables negotiation and usage of Comfort Noise CN 0 Disable default 1 Enable The use of CN is indicated by including a payload type for CN on the media description line of the SDP The device can use CN with a codec whose RTP time stamp clock rate is 8 000 Hz G 711 G 726 The static payload type 13 is used The use of CN is negotiated between sides Therefore if the remote side doesn t support CN it is not used Note Silence Suppression must be enabled to generate CN Determines the index of the fi
499. ol Privacy Protocol Authentication Key Privacy Key T None Read Write 3 To add an SNMP v3 user in the Add field enter the desired row index and then click Add A new row appears 4 Configure the SNMP V3 Setting parameters according to the table below 5 Click the Apply button to save your changes 6 To save the changes refer to Saving Configuration on page 161 For a description of the web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 34 You can also configure SNMP v3 users using the ini file table parameter SNMPUsers refer to SNMP Parameters on page 242 SIP User s Manual 156 Document LTRT 65413 SIP User s Manual Parameter Index SNMPUsers Index User Name SNMPUsers Username Authentication Protocol SNMPUsers AuthProtocol Privacy Protocol SNMPUsers PrivProtocol Authentication Key SNMPUsers AuthKey Privacy Key SNMPUsers PrivKey Group SNMPUsers Group 3 Web Based Management Table 3 28 SNMP V3 Users Parameters Description The table index The valid range is 0 to 9 Name of the SNMP v3 user This name must be unique Authentication protocol of the SNMP v3 user 0 None default 1 MD5 2 SHA 1 Privacy protocol of the SNMP v3 user 0 None default 1 DES 2 3DES 3 AES 128 4 AES 192 5 AES 256 Authentication key Keys can be entered in the for
500. old If a Resume un hold Re INVITE message is received before the timer expires the call is renewed If this timer expires the call is released 1 The call is placed on hold indefinitely until the initiator of on hold retrieves the call again default 0 2400 Time to wait in seconds after which the call is released Defines the timeout in seconds for applying the Call Hold Reminder Ring If a user hangs up while a call is still on hold then the FXS interface immediately rings the extension for the duration specified by this parameter If the user off hooks the phone the call becomes active The valid range is 0 to 600 The default value is 30 Note This parameter is applicable only to FXS interfaces 301 March 2010 ca AudioCodes 6 8 6 MediaPack Series Call Transfer Parameters The call transfer parameters are described in the table below Table 6 39 Call Transfer Parameters Parameter Web EMS Enable Transfer EnableTransfer Web Transfer Prefix EMS Logical Prefix For Transferred Call xferPrefix Web Transfer Prefix IP 2 Tel XferPrefixIP2Tel Web EMS Enable Semi Attended Transfer EnableSemiAttendedTransfer EMS Blind Transfer Add Prefix KeyBlindTransferAddPrefix SIP User s Manual Description Determines whether call transfer is enabled 0 Disable Disable the call transfer service 1 Enable Enable the call transfer service using REFER default If t
501. on Web Max Allowed Packet Loss for Alt Routing IPConnQoSMaxAllowedPL Web Max Allowed Delay for Alt Routing msec IPConnQoSMaxAllowedDelay Description Determines the duration in milliseconds for which the device plays a tone to the endpoint on each Alternative Routing attempt When the device finishes playing the tone a new SIP INVITE message is sent to the new destination The tone played is the Call Forward Tone Tone Type 25 in the CPT file The valid range is 0 to 20 000 The default is 0 i e no tone is played Packet loss in percentage at which the IP connection is considered a failure and Alternative Routing mechanism is activated The default value is 20 Transmission delay in msec at which the IP connection is considered a failure and the Alternative Routing mechanism is activated The range is 100 to 10 000 The default value is 250 Web Reasons for Alternative Tel to IP Routing Table EMS Alt Route Cause Tel to IP AltRouteCauseTel2IP Version 6 0 This ini file table parameter configures SIP call failure reason values received from the IP side If an IP call is released as a result of one of these reasons the device attempts to locate an alternative IP route address for the call in the Tel to IP Routing if a Proxy is not used or used as a redundant Proxy you need to set the parameter RedundantRoutingMode to 2 The release reason for Tel to IP calls is provided in SIP 4xx 5x
502. on request is resent according to the parameter RegistrationTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the device resends its registration request after 3600 x 70 2520 sec The default value of RegistrationTimeDivider is 50 If registration per channel is selected on device startup the device sends REGISTER requests according to the maximum number of allowed SIP dialogs configured by the parameter NumberOfActiveDialogs After each received response the subsequent REGISTER request is sent 9 8 4 Establishing a Call between Two Devices This section provides an example on configuring two AudioCodes devices with FXS interfaces for establishing call communication After configuration you can make calls between telephones connected to the same device and between the two devices SIP User s Manual 426 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities This example assumes the following m The IP address of the first device is 10 2 37 10 and its endpoint numbers are 101 to 104 m The IP address of the second device is 10 2 37 20 and its endpoint numbers are 201 to 204 m ASIP Proxy is not used Internal call routing is performed using the device s Tel to IP Routing gt To configure the two devices for call communication 1 For the first device 10 2 37 10 in the Endpoint Phone Number Table page refer to Configuring the
503. on page 24 SIP User s Manual 46 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 2 Using the Home Page The Home page provides you with a graphical display of the device s front panel displaying color coded status icons for monitoring the functioning of the device The Home page also displays general device information in the General Information pane such as the device s IP address and firmware version By default the Home page is displayed when you access the device s Web interface gt To access the Home page m On the toolbar click the Home icon the Home page is displayed Figure 3 26 MP 11x Home Page Uplink F ail Ready P ower Note The displayed number and type FXO and or FXS of channels depends on the device s model e g MP 118 or MP 114 The table below describes the areas of the Home page Table 3 3 Description of the Areas of the Home Page Label Description Alarms Displays the highest severity of an active alarm raised if any by the device Green no alarms Red Critical alarm Orange Major alarm Yellow Minor alarm To view a list of active alarms in the Active Alarms page refer to Viewing Active Alarms on page 176 click the Alarms area Version 6 0 47 March 2010 Ao wal AudioCodes MediaPack Series Channel Ports Uplink MP 11x LAN MP 124 3 2 1 Fail Ready Power Displays the status of the ports channels
504. on 6 0 137 March 2010 7a c tal AudioCodes MediaPack Series gt To configure Automatic Dialing 1 Open the Automatic Dialing page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Automatic Dialing page item Figure 3 85 Automatic Dialing Page Gateway Destination Phone Auto Dial Port Number Status Port 1 101 Enable Port 2 911 Hotline W Port 3 Enable V Port 4 Enable Port 5 Enable Port 6 FXO Enable Port 7 FXO Enable Port 8 FXO Enable V 2 In the Destination Phone Number field corresponding to a port enter the telephone number that you want automatically dialed 3 From the Auto Dial Status drop down list select one of the following e Enable 1 The number in the Destination Phone Number field is automatically dialed if the phone is off hooked for FXS interfaces or a ring signal from PBX PSTN switch is detected FXO interfaces The FXO line is seized only after the SIP call is answered e Disable 0 The automatic dialing feature for the specific port is disabled i e the number in the Destination Phone Number field is ignored e Hotline 2 FXS interfaces When a phone is off hooked and no digit is dialed for a user defined time configured using the parameter HotLineToneDuration the number in the Destination Phone Number field is automatically dialed
505. on IP address after the Tel to IP call is answered 0 Disable Disabled default 1 Enable Enable digit delivery to IP To enable this feature modify the called number to include at least one p character The device uses the digits before the p character in the initial INVITE message After the call is answered the device waits for the required time number of p multiplied by 1 5 seconds and then sends the rest of the DTMF digits using the method chosen in band or out of band Notes For this parameter to take effect a device reset is required The called number can include several p characters 1 5 seconds pause for example 1001pp699 8888p9p300 Enables the Digit Delivery feature which sends DTMF digits of the called number to the device s port phone line after the call is answered i e line is off hooked for FXS or seized for FXO for IP to Tel calls 0 Disable Disabled default 1 Enable Enable Digit Delivery feature for the FXO FXS device Notes For this parameter to take effect a device reset is required The called number can include characters p 1 5 seconds pause and d detection of dial tone If character d is used it must be the first digit in the called number The character p can be used several times For example for FXS FXO interfaces the called number can be as follows d1005 dpp699 p9p300 To add the d and p digits use the usual num
506. on Page w General Settings MAC Address oo908fOs4fo9 Serial Number 544665 Board Type MP 118 FAS _ FRO Device Up Time Od 0h 10m 35s 47th Device Administrative State Unlocked Device Operational State Enabled Flash Size bytes 8388606 RAM Size bytes 33554432 CPU Speed MHz 40 w Versions Version ID 6 004 002 011 DSP Type 0 DSP Software Version 60007 DSP Software Name 204IM Flash Version 199 wv Loaded Files Loaded Call Progress Tones Default Progress Tones Loaded Coder Table Default CODERTABLE gt To delete a loaded file m Click the Delete button corresponding to the file that you want to delete Deleting a file takes effect only after device reset refer to Resetting the Device on page 159 Version 6 0 175 March 2010 A K tal AudioCodes MediaPack Series 3 5 1 5 Viewing Performance Statistics The Performance Statistics page provides read only device performance statistics This page is refreshed with new statistics every 60 seconds The duration that the current statistics has been collected is displayed above the statistics table gt To view performance statistics m Open the Performance Statistics page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Performance Statistics page item Figure 3 112 Performa
507. on on RTP multiplexing refer to RTP Multiplexing ThroughPacket on page 440 Notes The value of this parameter on the local device must equal the value of BaseUDPPort on the remote device To enable RTP multiplexing the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort must be set to a non zero value When VLANs are implemented RTP multiplexing is not supported 359 March 2010 ca AudioCodes Parameter Web RTP Multiplexing Local UDP Port L1L1ComplexTxUDPPort Web RTP Multiplexing Remote UDP Port L1L1ComplexRxUDPPort EMS No Op Enable NoOpEnable EMS No Op Interval NoOplnterval EMS No Op Payload Type RTPNoOpPayloadType Web RTCP Packet Interval EMS Packet Interval RTCPInterval Web Disable RTCP Interval Randomization EMS Disable Interval Randomization DisableRTCPRandomize SIP User s Manual MediaPack Series Description Determines the local UDP port used for outgoing multiplexed RTP packets applies to RTP multiplexing The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 i e RTP multiplexing is disabled Note For this parameter to take effect a device reset is reguired Determines the remote UDP port to where the multiplexed RTP packets are sent and the local UDP port used for incoming multiplexed RTP packets applies to RTP multiplexing The valid range is the range of possible UDP ports 6 000 to 64 000 Th
508. on types in different setups These application types are configurable The applications listed below can be configured to one of two application types m DNS m NTP SIP User s Manual 454 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities Table 10 7 Application Type Parameters Parameter Description EnableDNSasOAM This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for DNS services VLAN Determines the traffic type for DNS services 1 OAMP default 0 Control Note For this parameter to take effect a device reset is required EnableNTPasOAM This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for NTP services VLAN Determines the traffic type for NTP services 1 OAMP default 0 Control Note For this parameter to take effect a device reset is required 10 8 1 4 Multiple Interface Table Configuration Summary and Guidelines Multiple Interface table configuration must adhere to the following rules Version 6 0 Up to 16 different interfaces may be defined The indices used must be in the range between 0 to 15 Each interface must have its own subnet Defining two interfaces with addresses in the same subnet i e two interfaces with 192 168 0 1 16 and 192 168 100 1 16 is illegal Subnets in different interfaces must not be overlapping in any way i e defining two
509. ons in this table becomes available when booting from flash again This enables the device to operate with a temporary address for initial management and configuration while retaining the address to be used for deployment Defines the Classless Inter Domain Routing CIDR style representation of a dotted decimal subnet notation The CIDR style representation uses a suffix indicating the number of bits which are set in the dotted decimal format e g 192 168 0 0 16 is synonymous with 192 168 0 0 and a subnet of 255 255 0 0 Defines the number of 1 bits in the subnet mask i e replaces the standard dotted decimal representation of the subnet mask for IPv4 interfaces For example A subnet mask of 255 0 0 0 is represented by a prefix length of 8 i e 11111111 00000000 00000000 00000000 and a subnet mask of 255 255 255 252 is represented by a prefix length of 30 i e 11111111 11111111 11111111 11111100 The prefix length is a Classless Inter Domain Routing CIDR style presentation of a dotted decimal subnet notation The CIDR style presentation is the latest method for interpretation of IP addresses Specifically instead of using eight bit address blocks it uses the variable length subnet masking technique to allow allocation on arbitrary length prefixes refer to http en wikipedia org wiki Classless Inter Domain Routing for more information For IPv4 Interfaces the prefix length values range from 0 to 31 Note Subnets o
510. operates in Automatic Dialing mode there is no method to inform the PBX if a Tel to IP call has failed SIP error response 4xx 5xx or 6xx is received The reason is that the FXO device does not seize the line until a SIP 200 OK response is received Use the FXOAutoDialPlayBusyTone parameter to allow the device to play a Busy Reorder tone to the PSTN line if a SIP error response is received The FXO device seizes the line off hook for the duration defined by the TimeForReorderTone parameter After playing the tone the line is released on hook m Call termination after caller PBX on hooks phone Ring Detection Timeout feature This method operates in one of the following manners e Automatic Dialing is enabled if the remote IP party doesn t answer the call and the ringing signal from the PBX stops for a user defined time configured by the parameter FXOBetweenRingTime the FXO device releases the IP call e No automatic dialing and Caller ID is enabled the device seizes the line after detection of the second ring signal allowing detection of caller ID sent between the first and the second rings If the second ring signal is not received within this timeout the device doesn t initiate a call to IP 9 4 2 3 3 Ring Detection Timeout The operation of Ring Detection Timeout depends on the following m Automatic dialing is disabled and Caller ID is enabled if the second ring signal is not received for a user defined time using t
511. ops sending RTP and plays a local Held tone m When an active call receives a Re INVITE message with the sendonly string in SDP the device stops sending RTP and listens to the remote party In this mode it is expected that on hold music or any other hold tone is played over IP by the remote party You can also configure the device to keep a call on hold for a user defined time after which the call is disconnected using the ini file parameter HeldTimeout The device also supports double call hold for FXS interfaces where the called party which has been placed on hold by the calling party can then place the calling party on hold as well and make a call to another destination The flowchart below provides an example of this type of call hold Figure 9 17 Double Hold SIP Call Flow Endpoint C Endpoint A Endpoint B Endpoint D l INVITE sendrecv l 200 OK sendrecv Conversation gt n w z 7 INVITE Hold inactive 200 OK inactive Conversation gt Ea INVITE sendrecv 200 OK sendrecv INVITE Hold inactive INVITE Retrieve sendrecv Conversation gt 200 OK inactive j 2000K inactive Z MMMM INVITE Retrieve sendrecv INVITE Hold inactive gt 1 200 OK sendrecv 1 200 OK inactive 1 r 1 I I I 1 I I I I I I I 1 1 1 gt t 1 1 Conversation gt 1 I I j SIP User s Manual 410 Document LTRT 65413 SIP User s Manual 9 IP Telephony Cap
512. or Frequency Defines the deviation in Hz allowed for the detection of each Deviation CPT signal frequency CPTDetectorFrequencyDeviation The valid range is 1 to 30 The default value is 10 Note For this parameter to take effect a device reset is required 6 11 3 Metering Tone Parameters The metering tone parameters are described in the table below Table 6 50 Metering Tone Parameters Parameter Description Web Generate Metering Determines the method used to configure the metering tones that are Tones generated to the Tel side EMS Metering Mode PayPhoneMeteringMode 0 Disable Metering tones aren t generated default 1 Internal Table Metering tones are generated according to the internal table configured by the parameter ChargeCode Notes This parameter is applicable only to FXS interfaces If you select Internal Table you must configure the Charge Codes Table refer to Configuring the Charge Codes Table on page 113 Web Analog Metering Determines the metering method for generating pulses sinusoidal Type metering burst frequency by the FXS port EMS Metering Type a e t MeteringType 0 12 KHz default 12 kHz sinusoidal bursts 1 16 KHz 16 kHz sinusoidal bursts 2 Polarity Reversal pulses Notes For this parameter to take effect a device reset is required This parameter is applicable only to FXS interfaces Web Analog TTX Voltage Determines the m
513. or multiple key S N lines e Single key S N line a Open the Software Upgrade Key text file using for example Microsoft Notepad b Select and copy the key string of the device s S N and paste it into the field Add a Software Upgrade Key c Click the Add Key button SIP User s Manual 166 Document LTRT 65413 SIP User s Manual 3 Web Based Management e Multiple S N lines as shown below Figure 3 106 Software Upgrade Key with Multiple S N Lines sampleri Notepad OF WMPDE yensoix4PbBF 8eOZ4by ASa5h64 1R1aOksEb9AddF 89385 KeTIAddFSc1ss O2x1aOkeTJIAdGF 8c ts TJANgSaSh fy1aOkexZAddF8ahss a inthe Send Upgrade Key file field click the Browse button and navigate to the folder in which the Software Upgrade Key text file is located on your PC b Click the Send File button the new key is loaded to the device and validated If the key is valid it is burned to memory and displayed in the Current Key field 5 Verify that the Software Upgrade Key file was successfully loaded to the device by using one of the following methods e Inthe Key features group ensure that the features and capabilities activated by the installed string match those that were ordered e Access the Syslog server refer to the Product Reference Manual and ensure that the following message appears in the Syslog server S N___ Key Was Updated The Board Needs to be Reloaded with ini file n 6 Reset the d
514. or playing the special dial tone the received SIP NOTIFY message must contain the following headers m From and To contain the same information indicating the specific endpoint m Event ua profile m Content Type application simservs xml E Message body is the XML body and contains the dial tone pattern set to special condition tone lt ss dial tone pattern gt special condition tone lt ss dial tone pattern gt which is the special tone indication For cancelling the special dial tone and playing the regular dial tone the received SIP NOTIFY message must contain the following headers m From and To contain the same information indicating the specific endpoint m Event ua profile m Content Type application simservs xml m Message body is the XML body containing the dial tone pattern set to standard condition tone lt ss dial tone pattern gt standard condition tone lt ss dial tone pattern gt which is the regular dial tone indication Therefore the special dial tone is valid until another SIP NOTIFY is received that instructs otherwise as described above Note if the MWI service is active the MWI dial tone overrides this special Call Forward dial tone 9 7 6 Call Waiting The Call Waiting feature enables FXS devices to accept an additional second call on busy endpoints If an incoming IP call is designated to a busy port the called party hears a call waiting tone several configurable sho
515. or that Profile If the Profile is set to some limit the device maintains the number of concurrent calls incoming and outgoing pertaining to the specific Profile A limit value of 1 indicates that there is no limitation on calls default A limit value of 0 indicates that all calls are rejected When the number of concurrent calls is equal to the limit the device rejects any new incoming and outgoing calls pertaining to that profile RxDTMFOption configures the received DTMF negotiation method 1 not configured use the global parameter 0 don t declare RFC 287 March 2010 A tal AudioCodes MediaPack Series Parameter Description 2833 1 declare RFC 2833 payload type is SDP FirstTxDtmfOption and SecondTxDtmfOption configures the transmit DTMF negotiation method 1 not configured use the global parameter for the remaining options refer to the global parameter IP Profiles can also be used when operating with a Proxy server set the parameter AlwaysUseRouteTable to 1 Fora detailed description of each parameter refer to its corresponding global parameter For a description of using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Tel Profile Settings Table EMS Protocol Definition gt Telephony Profile TelProfile SIP User s Manual This ini file table parameter configures the Tel Profile table Each Tel Profile ID includes a set of par
516. ording to the following rules m The servername is equal to RegistrarName if configured The RegistrarName can be any string m Otherwise the servername is equal to RegistrarlP either FQDN or numerical IP address if configured m Otherwise the servername is equal to ProxyName if configured The ProxyName can be any string m Otherwise the servername is equal to ProxylP either FQDN or numerical IP address The parameter GWRegistrationName can be any string This parameter is used only if registration is per device If the parameter is not defined the parameter UserName is used instead If the registration is per endpoint the endpoint phone number is used The sipgatewayname parameter defined in the ini file or Web interface can be any string Some Proxy servers require that the sipgatewayname in REGISTER messages is set equal to the Registrar Proxy IP address or to the Registrar Proxy domain name The sipgatewayname parameter can be overwritten by the TrunkGroupSettings GatewayName value if the TrunkGroupSettings RegistrationMode is set to Per Endpoint REGISTER messages are sent to the Registrar s IP address if configured or to the Proxy s IP address A single message is sent once per device or messages are sent per channel according to the parameter AuthenticationMode There is also an option to configure registration mode per Hunt Group using the TrunkGroupSettings table The registrati
517. ority Media Premium Priority Control Premium Priority Gold Priority Bronze Priority w Differential Services Network QoS Media Premium QoS Control Premium QoS Gold QoS Bronze QoS 2 Configure the QoS parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 2 Media Settings The Media Settings menu allows you to configure the device s channel parameters This menu contains the following items m Voice Settings refer to Configuring the Voice Settings on page 61 m Fax Modem CID Settings refer to Configuring the Fax Modem CID Settings on page 61 RTP RTCP Settings refer to Configuring the RTP RTCP Settings on page 63 General Media Settings refer to Configuring the General Media Settings on page 64 Analog Settings refer to Configuring the Analog Settings on page 64 Media Security refer to Configuring Media Security on page 65 Channel parameters can be modified on the fly Changes take effect from the next call Some channel parameters can be configured per channel or call routing using profiles refer to Coders and Profile Definitions on page 101 SIP User s Manual 60 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 2 1 Configuring the Voice Settings The Voice Settings page is used for configuring various voice pa
518. oting the m Table ee Oe 10 8 3 Setting up the Device n 2m 10 8 3 1 Using the Web Int face 0 10 8 3 2 Using the ini File Nici okn Kdo na 11 SIF Sofware Packa Bzu k k 465 12 Selected Technical Specifications cscsscseesssssessseeceeeeeeeeeeeeeeeeeeeeeees 407 SIP User s Manual 8 Document LTRT 65413 List of Figures Figure 1 1 T Figure 3 Figure 3 Figure 3 Figur nfirma Box for Exiting Scenario Mode Fi gure 3 quest Page PEIE AEE AAA AENA ed Certificate Files ee 7a 4 e AudioCodes MediaPack Series Figure 3 56 Figure 3 57 Figure 3 58 Figure 3 59 Figure 3 60 Figure 3 61 Figure 3 62 Figure 3 63 Figure 3 64 Figure 3 65 Figure 3 66 Figure 3 67 Figure 3 68 69 Sup Figure 3 70 Figure 3 71 Figure 3 72 Key Figure 3 73 General Sett Figure 3 74 Figure 3 75 Figure 3 76 Figure 3 77 Figure 3 78 Figure 3 79 Figure 3 80 Figure 3 81 Figure 3 82 Figure 3 83 Figure 3 84 Figure 3 85 Figure 3 86 Figure 3 87 Figure 3 88 Figure 3 89 Figure 3 90 Figure 3 91 Figure 3 92 Figure 3 93 Figure Figure 3 94 Figure 3 95 Figure 3 96 SI Figure 3 97 Figure 3 98 Figure 3 99 SN Figure 3 100 Figure 3 101 Figure 3 102 Figure 3 103 Device Lock Confirmation me B Figure 3 104 Load Auxiliary Files Page is Figure 3 105 Software Upgrade Key Page Figure 3 106 Figure 3 107 Appl
519. ou can also access the IP2IP Routing Table page for configuring SAS routing rules refer to Configuring the IP2IP Routing Table SAS on page 146 The SAS menu and its page items appear only if you have enabled the SAS application refer to Enabling Applications on page 84 and the SAS application is included in the device s Software Upgrade Key refer to Loading a Software Upgrade Key on page 165 For a detailed explanation on SAS refer to Stand Alone Survivability SAS Feature on page 381 SIP User s Manual 144 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 11 1 Configuring Stand Alone Survivability Parameters The SAS Configuration page allows you to configure the device s Stand Alone Survivability SAS feature This feature is useful for providing a local backup through the PSTN in Small or Medium Enterprises SME that are serviced by IP Centrex services In such environments the enterprise s incoming and outgoing telephone calls external and internal are controlled by the Proxy which communicates with the enterprise through the WAN interface SAS ensures that incoming outgoing and internal calls service is maintained in case of WAN or Proxy failure using a PSTN or an alternative VoIP backup connection and the device s internal call routing To utilize the SAS feature the VoIP CPEs such as IP phones or residential gateways need to be defined so that their Proxy and Registrar d
520. ou make throughout the Web interface s pages are temporarily saved to the volatile memory RAM when you click the Submit button on these pages Parameter settings that are only saved to the device s RAM revert to their previous settings after a hardware software reset or power failure Therefore to ensure that your configuration changes are retained you must save them to the device s flash memory using the burn option described below To save the changes to the non volatile flash memory 1 Open the Maintenance Actions page refer to Maintenance Actions on page 159 2 Under the Save Configuration group click the BURN button a confirmation message appears when the configuration successfully saves Saving configuration to the non volatile memory may disrupt current traffic on the device To avoid this disable all new traffic before saving by performing a graceful lock refer to Locking and Unlocking the Device on page 161 Throughout the Web interface parameters preceded by the lightning symbol are not applied on the fly and require that you reset the device for them to take effect refer to Resetting the Device on page 159 SIP User s Manual 162 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 4 2 Software Update The Software Update menu allows you to upgrade the device s software by loading a new cmp file compressed firmware along with the ini file and a suite of auxiliary files
521. ould be configured to five seconds i e greater than the off time e g four Determines the number of rings before the device starts detecting Caller ID 0 0 Before first ring 1 1 After first ring default 2 2 After second ring Note This parameter is applicable only to FXO interfaces 329 March 2010 K tal AudioCodes MediaPack Series Parameter Description Web EMS Guard Time Defines the time interval in seconds after a call has ended and a new Between Calls call can be accepted for IP to Tel FXO calls GuardTimeBetweenCalls The valid range is 0 to 10 The default value is 1 Notes Occasionally after a call ends and on hook is applied a delay is required before placing a new call and performing off hook This is necessary to prevent incorrect hook flash detection or other glare phenomena This parameter is applicable only to FXO interfaces 6 14 FXS Parameters The general FXS parameters are described in the table below Table 6 53 General FXS Parameters Parameter Description Web FXS Coefficient Type Determines the FXS line characteristics AC and DC according to USA EMS Country Coefficients or Europe TBR21 standards FXSCountryCoefficients 66 Europe TBR21 70 USA United States default Note For this parameter to take effect a device reset is required SIP User s Manual 330 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters
522. outine 1 a RTP DSCP for MLPP Priority 1 RTP DSCP for MLPP Immediate 1 RTP DSCP for MLPP Flash 1 a a HK RTP DSCP for MLPP Flash Override 1 a RTP DSCP for MLPP Flash Override Override 1 a E911 MLPP Behavior standardMode 3 Configure the MLPP parameters as required Note If the following RTP DSCP parameters are set to 1 i e Not Configured Default the DiffServ value is set with the PremiumServiceClassMediaDiffserv global gateway parameter or by using IP Profiles MLPPRoutineRTPDSCP MLPPPriorityRTPDSCP MLPPlmmediateRTPDSCP MLPPFlashRTPDSCP MLPPFlashOverRTPDSCP MLPPFlashOverOverRTPDSCP MLPPNormalizedServiceDomain 5 8 Configuring the Device to Operate with SNMPv3 This section describes the SNMPv3 configuration process E Configuring SNMPv3 using SSH mE Configuring SNMPv3 using EMS non configured SNMPv3 System E Configuring SNMPv3 using EMS pre configured SNMPv3 System SIP User s Manual 200 Document LTRT 65413 SIP User s Manual 5 Element Management System EMS Note After configuring SNMPv3 ensure that you disable IPSec 5 8 1 Configuring SNMPv3 using SSH The procedure below describes how to configure SNMPv3 using SSH gt 1 Version 6 0 To configure the device to operate with SNMPv3 via SSH Open an SSH Client session e g PuTTY and then connect using the default user name and password Admin case sensitive to the device If a
523. p Payload Format for RTP This IETF document defines a No Op payload format for RTP The draft defines the RTP payload type as dynamic You can control the payload type with which the No Op packets are sent This is performed using the RTPNoOpPayloadType ini parameter refer to Networking Parameters on page 207 AudioCodes default payload type is 120 m T 38 No Op T 38 No Op packets are sent only while a T 38 session is activated Sent packets are a duplication of the previously sent frame including duplication of the sequence number Note Receipt of No Op packets is always supported IP Multicasting The device supports IP Multicasting level 1 according to RFC 2236 i e IGMP version 2 for RTP channels The device is capable of transmitting and receiving Multicast packets Robust Receipt of Media Streams This mechanism filters out unwanted RTP streams that are sent to the same port number on the device These multiple RTP streams can result from traces of previous calls call control errors and deliberate attacks When more than one RTP stream reaches the device on the same port number the device accepts only one of the RTP streams and rejects the rest of the streams The RTP stream is selected according to the following The first packet arriving on a newly opened channel sets the source IP address and UDP port from which further packets are received Thus the source IP address and UDP port identify the currently accept
524. p to which the incoming SIP call is assigned if it matches all or any combination of the parameters described above The IP Profile configured in Configuring P Profiles on page 107 to assign to the IP to Tel call The source IP Group associated with the incoming IP to Tel call This is the IP Group from where the INVITE message originated This IP Group can later be used as the Serving IP Group in the Account table for obtaining authentication user name password for this call refer to Configuring the Account Table on page 93 133 March 2010 7a L tal AudioCodes MediaPack Series 3 3 4 8 5 Configuring the Internal DNS Table The Internal DNS Table page similar to a DNS resolution is used to translate up to 20 host domain names into IP addresses e g when using the Tel to IP Routing Up to four different IP addresses can be assigned to the same host name typically used for alternative routing for Tel to IP call routing The device initially attempts to resolve a domain name using the Internal DNS table If the domain name isn t listed in the table the device performs a DNS resolution using an external DNS server You can also configure the DNS table using the ini file table parameter DNS2IP refer to DNS Parameters on page 218 gt To configure the internal DNS table 1 Open the Internal DNS Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Internal
525. page 93 m Proxy amp Registration refer to Configuring Proxy and Registration Parameters on page 96 m Proxy Sets Table refer to Configuring the Proxy Sets Table on page 97 SIP User s Manual 90 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 4 4 1 Configuring the IP Groups The IP Group Table page allows you to create up to nine logical IP entities called P Groups These IP Groups are used for call routing The IP Group can be used as a destination entity in the Tel to IP Routing and as a Serving IP Group in the Hunt Group Settings refer to Configuring Hunt Group Settings on page 85 and Account refer to Configuring the Account Table on page 93 tables These call routing tables are used for identifying the IP Group from where the INVITE is sent for obtaining a digest user password from the Account table if there is a need to authenticate subsequent SIP requests in the call The IP Group can also be implemented in IP to Tel call routing as a source IP Group The IP Groups can be assigned various entities such as a Proxy Set ID which represents an IP address created in Configuring the Proxy Sets Table on page 97 You can also assign the IP Group with a host name and other parameters that reflect parameters sent in the SIP Request From To headers When working with multiple IP Groups the default Proxy server should not be used i e the parameter IsProxyUsed must be set to 0 You can als
526. parameter TimeBetweenDigits This is the time that the device waits between each received digit When this inter digit timeout expires the device uses the collected digits to dial the called destination number m Pound key is pressed m Digit map pattern is matched Digit map pattern rules are defined by the parameter DigitMapping If the digit string i e dialed number matches one of the patterns in the digit map the device stops collecting digits and establishes a call with the collected number The digit map pattern can contain up to 52 options rules each separated by a vertical bar The maximum length of the entire digit pattern is 152 characters The available notations are described in the table below Table 9 2 Digit Map Pattern Notations Notation Description n m Range of numbers not letters single dot Repeat digits until next notation e g T Any single digit Dial timeout configured by the parameter TimeBetweenDigits Immediately applies a specific rule that is part of a general rule For example if a digit map includes a general rule x T and a specific rule 11x for the specific rule to take precedence over the general rule append S to the specific rule i e 11xS Below is an example of a digit map pattern containing eight rules DigitMapping 11xS 00 1 7 xxx 8xxxxxxx HXXXXXXX xx 91XXXXXXXXXX 9011x x T In the example above the rule 00 1 7 xxx denotes di
527. parameter to take effect a device reset is required Secret used to authenticate the device to the RADIUS server This should be a cryptically strong password Defines the default access level for the device when the RADIUS authentication response doesn t include an access level attribute The valid range is 0 to 255 The default value is 200 Security Administrator Defines the device s mode of operation regarding the timer configured by the parameter RadiusLocalCacheTimeout that determines the validity of the user name and password verified by the RADIUS server 0 Absolute Expiry Timer when you access a Web page the timeout doesn t reset instead it continues decreasing 1 Reset Timer Upon Access upon each access to a Web page the timeout always resets reverts to the initial value configured by RadiusLocalCacheTimeout Defines the time in seconds the locally stored user name and password verified by the RADIUS server are valid When this time expires the user name and password become invalid and a must be re verified with the RADIUS server The valid range is 1 to OxFFFFFF The default value is 300 5 minutes 1 Never expires 0 Each request requires RADIUS authentication Defines the vendor ID that the device accepts when parsing a RADIUS response packet The valid range is 0 to OxFFFFFFFF The default value is 5003 Defines the code that indicates the access level attribute i
528. perating with multiple Internet Telephony Service Providers ITSP for VoIP services The device supports the SIP protocol enabling the deployment of VoIP solutions in environments where each enterprise or residential location is provided with a simple media gateway This provides the enterprise with a telephone connection i e RJ 11 connector and the capability to transmit voice and telephony signals over a packet network The device provides FXO and or FXS analog ports for direct connection to an enterprise s PBX FXO and or to phones fax machines and modems FXS Depending on model the device can support up to 24 simultaneous VoIP calls The device is also equipped with a 10 100Base TX Ethernet port for connection to the IP network The device provides LEDs for indicating operating status of the various interfaces The device is a compact unit that can be easily mounted on a desktop wall or in a 19 inch rack The device provides a variety of management and provisioning tools including an HTTP based embedded Web server Telnet Element Management System EMS and Simple Network Management Protocol SNMP The user friendly Web interface provides remote configuration using any standard Web browser such as Microsoft Internet Explorer Version 6 0 17 March 2010 7a E ll AudioCodes MediaPack Series The figure below illustrates a typical MediaPack VoIP application Figure 1 1 Typical MediaPack VoIP Application
529. port Caller ID generation for FXS interfaces and detection for FXO interfaces The format of this parameter is as follows EnableCallerlD FORMAT EnableCalleriD Index EnableCallerlD IsEnabled EnableCallerlD Where Index Port number where 0 depicts Port 1 IsEnabled v 0 Disable disables Caller ID default v 1 Enable enables Caller ID generation FXS or detection FXO For example EnableCallerlD 0 EnableCallerlD 1 Notes 1 caller ID enabled on Port 1 0 caller ID disabled on Port 2 Ifa port is not configured its Caller ID generation detection is determined according to the global parameter EnableCallerlD For configuring this table using the Web interface refer to Configuring Caller ID Permissions on page 141 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Caller Display Information Table EMS SIP Endpoints gt Caller ID CallerDisplaylnfo This ini file table parameter enables the device to send Caller ID information to IP when a call is made The called party can use this information for caller identification The information Version 6 0 289 March 2010 ca AudioCodes Parameter SIP User s Manual MediaPack Series Description configured in this table is sent in the SIP INVITE message s From header The format of this parameter is as follows CallerDisplayInfo FORMAT CallerDisplaylnfo Ind
530. port mode used by the device 0 Disable transparent mode 1 T 38 Relay default 2 Bypass 3 Events Only Note This parameter is overridden by the parameter IsFaxUsed If the parameter IsFaxUsed is set to 1 T 38 Relay or 3 Fax Fallback then FaxTransportMode is always set to 1 T 38 relay Number of times that control packets are retransmitted when using the T 38 standard The valid range is 0 to 4 The default value is 2 Number of times that each fax relay payload is retransmitted to the network 0 No redundancy default 1 One packet redundancy 2 Two packet redundancy Note This parameter is applicable only to non V 21 packets Maximum rate in bps at which fax relay messages are transmitted outgoing calls 0 2400 2 4 kbps 1 4800 4 8 kbps 2 7200 7 2 kbps 3 9600 9 6 kbps 4 12000 12 0 kbps 5 14400 14 4 kbps default 6 16800bps 16 8 kbps 7 19200bps 19 2 kbps 8 21600bps 21 6 kbps 9 24000bps 24 kbps 10 26400bps 26 4 kbps 11 28800bps 28 8 kbps 12 31200bps 31 2 kbps 13 33600bps 33 6 kbps 351 March 2010 ca AudioCodes Parameter Web Fax Relay ECM Enable EMS Relay ECM Enable FaxRelayECMEnable Web Fax Modem Bypass Coder Type EMS Coder Type FaxModemBypassCoderType Web EMS CNG Detector Mode CNGDetectorMode Web Fax Modem Bypass Packing Factor EMS Pack
531. port range is the Base UDP Port 10 number of the device s channels The range of possible UDP ports is 6 000 to 64 000 The default base UDP port is 6000 For example if the Base UDP Port is set to 6000 then 1 one channel may use the ports RTP 6000 RTCP 6001 and T 38 6002 2 another channel may use RTP 6010 RTCP 6011 and T 38 6012 etc The UDP port range is as follows MP 112 MP 114 BaseUDPport to BaseUDPport 3 10 MP 118 BaseUDPport to BaseUDPport 7 10 MP 124 BaseUDPport to BaseUDPport 23 10 Notes For this parameter to take effect a device reset is reguired The UDP ports are allocated randomly to channels You can define a UDP port range per Media Realm refer to Configuring Media Realms If RTP Base UDP Port is not a factor of 10 the following message is generated invalid local RTP port For detailed information on the default RTP RTCP T 38 port allocation refer to the Product Reference Manual Determines the lower boundary of UDP ports used for RTP RTCP and T 38 by a remote device If this parameter is set to a non zero value ThroughPacket RTP multiplexing is enabled The device uses this parameter and BaseUDPPort to identify and distribute the payloads from the received multiplexed IP packet to the relevant channels The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 i e RTP multiplexing is disabled For detailed informati
532. pplicable only if the parameter RTPRedundancyDepth is set to 1 N A Controlled internally by the device according to the selected coder N A Controlled internally by the device according to the selected coder N A Controlled internally by the device according to the selected coder N A Use the ini file parameter RFC2833PayloadType instead N A Use the ini file parameter RFC2833PayloadType instead Changes the RTP packets according to the MAC address of received RTP packets and according to Gratuitous Address Resolution Protocol GARP messages 0 Nothing is changed 1 If the device receives RTP packets with a different source MAC address than the MAC address of the transmitted RTP packets then it sends RTP packets to this MAC address and removes this IP entry from the device s ARP cache table 2 The device uses the received GARP packets to 358 Document LTRT 65413 SIP User s Manual Parameter Web RTP Base UDP Port EMS Base UDP Port BaseUDPport Web Remote RTP Base UDP Port EMS Remote Base UDP Port RemoteBaseUDPPort Version 6 0 6 Configuration Parameters Reference Description change the MAC address of the transmitted RTP packets default 3 Options 1 and 2 are used Note For this parameter to take effect a device reset is reguired Lower boundary of the UDP port used for RTP RTCP RTP port 1 and T 38 RTP port 2 The upper boundary of the UDP
533. pplies Standard IP protocol numbers as defined by the Internet Assigned Numbers Authority IANA should be used for example 0 Any protocol default 17 UDP 6 TCP Determines the duration in seconds for which the negotiated IKE SA Main mode is valid After this time expires the SA is re negotiated Note Main mode negotiation is a processor intensive operation for best performance do not set this parameter to less than 28 800 i e eight hours The default value is 0 i e unlimited Determines the duration in seconds for which the negotiated IPSec SA Quick mode is valid After this time expires the SA is re negotiated The default value is 0 i e unlimited Note For best performance a value of 3 600 i e one hour or more is recommended Determines the maximum volume of traffic in kilobytes for which the negotiated IPSec SA Quick mode is valid After this specified volume is reached the SA is re negotiated The default value is 0 i e the value is ignored Configures dead peer detection DPD according to RFC 3706 0 DPD Disabled default 1 DPD Periodic DPD is enabled with message exchanges at regular intervals 2 DPD on demand DPD is enabled with on demand checks message exchanges as needed i e before sending data to the peer If the liveliness of the peer is questionable the device sends a DPD message to query the status of the peer If the device has no tr
534. pulation tables x n ly where m x any number of characters digits to add at the beginning of the number i e first digits in the prefix m n defines the location in the original destination or source number where the digits y are added e n location number of digits counted from the left of the number of a specific string in the original destination or source number e number of digits that this string includes m y prefix to add at the specified location For example assume that you want to manipulate an incoming IP call with destination number 5492028888888 area code 202 and phone number 8888888 to the number 0202158888888 To perform such a manipulation the following configuration is required in the Number Manipulation table 1 The following notation is used in the Prefix to Add field 0 5 3 15 where e Os the number to add at the beginning of the original destination number e 5 3 denotes a string that is located after and including the fifth character i e the first 2 in the example of the original destination number and its length being three digits i e the area code 202 in the example e 15 is the number to add immediately after the string denoted by 5 3 in other words 15 is added after i e to the right of the digits 202 2 The first seven digits from the left are removed from the original number by entering 7 in the Stripped Digits From Left field Figure
535. r it is reset The cmp file is validated before it s burned to flash The checksum of the cmp file is also compared to the previously burnt checksum to avoid unnecessary resets The maximum length of the URL address is 255 characters IniFileURL Specifies the name of the ini file and the path to the server IP address or FQDN on which it is located The ini file can be loaded using HTTP HTTPS FTP FTPS or NFS For example http 192 168 0 1 filename http 192 8 77 13 config lt MAC gt https lt username gt lt password gt A lt IP address gt lt file name gt Notes For this parameter to take effect a device reset is required When using HTTP or HTTPS the date and time of the ini file are validated Only more recently dated ini files are loaded The optional string lt MAC gt is replaced with the device s MAC address Therefore the device requests an ini file name that contains its MAC address This option allows the loading of specific configurations for specific devices The maximum length of the URL address is 99 characters PrtFileURL Specifies the name of the Prerecorded Tones file and the path to the server IP address or FQDN on which it is located For example http server_nameffile https server_nameffile Note The maximum length of the URL address is 99 characters CptFileURL Specifies the name of the CPT file and the path to the server IP address or FQDN on which it is loc
536. rameter of an INFO message plus the value of this parameter minus 1 The valid range is 1 to 1 000 The default value is 1 i e not used Notes Itis assumed that all Call Waiting Tones are defined in sequence in the CPT file SIP Alert Info header examples v Alert Info lt Bellcore dr2 gt v Alert Info lt http Bellcore dr2 gt where dr2 defines call waiting tone 2 The SIP INFO message is according to Broadsoft s application server definition Below is an example of such an INFO message INFO sip 06 192 168 13 2 5060 SIP 2 0 Via SIP 2 0 UDP 192 168 13 40 5060 branch z9hG4bK040066422630 From lt sip 4505656002 192 168 13 40 5060 gt tag 1455352915 To lt sip 06 192 168 13 2 5060 gt Call ID 0010 0008 192 168 13 2 CSeq 342168303 INFO Content Length 28 296 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Content Type application broadsoft play tone CallWaitingTone1 6 8 3 Call Forwarding Parameters The call forwarding parameters are described in the table below Table 6 36 Call Forwarding Parameters Parameter Description Web Enable Call Forward Determines whether Call Forward is enabled EnableForward 0 Disable Disable the Call Forward service 1 Enable Enable Call Forward service using REFER default For FXS interfaces the Call Forward table FwdInfo parameter must be defined to use the Call Forward service
537. rameters Files CallProgressTonesFileName cpusa dat 4 1 2 Configuring ini File Table Parameters The ini file table parameters allow you to configure tables which can include multiple parameters co umns and row entries index When loading an ini file to the device it s recommended to include only tables that belong to applications that are to be configured dynamic tables of other applications are empty but static tables are not The ini file table parameter is composed of the following elements m Title of the table The name of the table in square brackets e g MY_TABLE_NAME m Format line Specifies the columns of the table by their string names that are to be configured e The first word of the Format line must be FORMAT followed by the Index field name and then an equal sign After the equal sign the names of the columns are listed e Columns must be separated by a comma e The Format line must only include columns that can be modified i e parameters that are not specified as read only An exception is Index fields which are mandatory e The Format line must end with a semicolon m Data line s Contain the actual values of the columns parameters The values are interpreted according to the Format line e The first word of the Data line must be the table s string name followed by the Index field e Columns must be separated by a comma e A Data line must
538. rameters Open the Supplementary Services page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Supplementary Services page item Figure 3 69 Supplementary Services Page m Enable Hold Hold Format Held Timeout Call Hold Reminder Ring Timeout Enable Transfer Transfer Prefix Enable Call Forward Enable Call Waiting Number of Call Waiting Indications Time Between Call Waiting Indications Time Before Waiting Indications Waiting Beep Duration Enable Caller ID Hook Flash Code Flash Keys Seguence Style Flash Keys Seguence Timeout Caller ID Type Enable NRT Subscription 4S Subscribe IPGroupID NRT Subscribe Retry Time Call Forward Ring Tone ID Enable 0 0 0 0 1 30 Enable Enable Enable 2 10 0 300 Disable 0 2000 Standard Bellcore Disable 1 120 1 wv Message Waiting Indication MWI Parameters Enable MWI MWI Analog Lamp MWI Display Subscribe to MWI MWI Server IP Address MWI Server Transport Type MWI Subscribe Expiration Time Stutter Tone Duration MWI Subscribe Retry Time Disable Disable Disable No Not Configured 7200 2000 120 v Conference 6 Enable 3 Way Conference Establish Conference Code
539. rameters on page 186 This ini file table parameter configures the Proxy Set ID table It is used in conjunction with the ini file table parameter ProxylP which defines the Proxy Set IDs with their IP addresses The ProxySet ini file table parameter defines additional attributes per Proxy Set ID This includes for example Proxy keep alive and load balancing and redundancy mechanisms if a Proxy Set contains more than one proxy address The format of this parameter is as follows ProxySet FORMAT ProxySet Index ProxySet EnableProxyKeepAlive ProxySet ProxyKeepAliveTime ProxySet ProxyLoadBalancingMethod ProxySet IsProxyHotSwap ProxySet SRD ProxySet For example ProxySet 0 0 60 0 60 1 0 0 ProxySet 1 1 0 1 Notes This table parameter can include up to 10 indices 0 9 For configuring the Proxy Set IDs and their IP addresses 270 Document LTRT 65413 SIP User s Manual Parameter Registrar Parameters Web Enable Registration EMS Is Register Needed IsRegisterNeeded Web EMS Registrar Name RegistrarName Web Registrar IP Address EMS Registrar IP RegistrarlP Version 6 0 6 Configuration Parameters Reference Description use the parameter ProxylP The parameter ProxySet SRD is not applicable For configuring the Proxy Set ID table using the Web interface and for a detailed description of the parameters of this ini file table refer to Configuring the Proxy Sets
540. rameters such as voice volume silence suppression and DTMF transport type For a detailed description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt To configure the Voice parameters 1 Open the Voice Settings page Configuration tab gt Media Settings menu gt Voice Settings page item Figure 3 40 Voic e Settings Page vw Silence Suppression DTMF Transport Type DTMF Volume 31 to 0 NTE Max Duration Enable Answer Detector Answer Detector Sensiti 6 DTMF Generation Twist Echo Canceller Voice Volume 32 to 31 dB Input Gain 32 to 31 dB dB Answer Detector Activity Delay Answer Detector Silence Time Answer Detector Redirection vity 0 0 Disable RFC2833 Relay DTMF 11 1 Disable 0 10 0 0 0 Enable 2 Configure the Voice parameters as required 3 Click the Submit button to save your changes 4 Version 6 0 61 To save the changes to flash memory refer to Saving Configuration on page 161 March 2010 7a tal AudioCodes MediaPack Series 3 3 2 2 Configuring the Fax Modem CID Settings The Fax Modem CID Settings page is used for configuring fax modem and Caller ID CID parameters For a detailed description of the parameters appearing on this page refer to Configuration Parameter
541. rch 2010 A c tal AudioCodes MediaPack Series 3 3 4 9 6 Configuring Call Waiting The Call Waiting page allows you to enable or disable call waiting per device FXS port This page is applicable only to FXS interfaces Instead of using this page you can enable or disable call waiting for all the device s ports using the global call waiting parameter Enable Call Waiting refer to Configuring Supplementary Services on page 111 You can also configure the Call Waiting table using the ini file table parameter CallWaitingPerPort refer to SIP Configuration Parameters on page 245 For additional call waiting configuration refer to the following parameters FirstCallWaitingTonelD in the CPT file TimeBeforeWaitingIndication WaitingBeepDuration TimeBetweenWaitinglndications and NumberOfWaitingIndications gt To configure Call Waiting 1 Open the Caller Waiting page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Call Waiting page item Figure 3 89 Call Waiting Page Gateway Port FRS FRS FRS FRS FXO 2 From the Call Waiting Configuration drop down list corresponding to the port you want to configure for call waiting select one of the following options e Enable Enables call waiting for the specific port When the device receives a call on a busy endpoint port it responds with a 182 response not with a 486 busy The device plays a call wa
542. rect manipulation is performed only after the parameter CopyDest2RedirectNumber gt To configure the redirect Tel to IP table 1 Open the Redirect Number Tel gt IP page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt Redirect Number Tel gt IP page item Figure 3 75 Redirect Number Tel to IP Page Stripped Stripped Digits Digits From From Left Right Source Source Index Trunk IP Destination Prefix Redirect Prefix Prefix to Add Group Group Lok p F 555 3 o E sum to Add Number of Digits to Presentation Leave 5 Not Configured The figure below shows an example configuration in which the redirect prefix 555 is manipulated According to the configured rule if for example the number 5551234 is received after manipulation the device sends the number to IP as 91234 2 Configure the redirect number Tel to IP rules according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 SIP User s Manual 120 Document LTRT 65413 SIP User s Manual 3 Web Based Management Table 3 19 Redirect Number Tel to IP Parameters Description Parameter Source Trunk Group Web EMS Destination Prefix Web EMS Redirect Prefix Web Stripped Digits From Left EMS Remove From Left Web Stripped Digits From Right EMS Remove From Right
543. res accessibility to all parameters when creating a Scenario For a description on the Navigation tree views refer to Navigation Tree on page 27 If you previously created a Scenario and you click the Create Scenario button the previously created Scenario is deleted and replaced with the one you are creating Only users with access level of Security Administrator can create a Scenario SIP User s Manual 38 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 8 2 Accessing a Scenario Once you have created the Scenario you can access it at anytime by following the procedure below gt To access the Scenario 1 On the Navigation bar select the Scenario tab a message box appears requesting you to confirm the loading of the Scenario Figure 3 16 Scenario Loading Message Box Microsoft Internet Explorer A Loading Scenario PBX Interoperability 2 Click OK the Scenario and its Steps appear in the Navigation tree as shown in the example figure below Figure 3 17 Scenario Example Available Pa er Basic Parametar Ust a Max Digits In Phone Num 5 Scenario Name PBX Inter Digit Timeout for Overlap Dialing sec Interoperability Declare RFC 2833 in SOP i Define Coders Ist Tx DTMF Option Define Max Digits 2nd Tx OTMF Option Definie Voice Mail 3rd Tx DTMF Option 4th Tx OTMF Option Sth Tx DTMF Option RFC 2633 Payload Type Hook Flash Option 4 Digit Mapping Rules Dial Tone Dur
544. ress regardless of the address specified in the Multiple Interface table This configured address becomes available when booting from flash m Network Configuration changes are offline The new configuration should be saved and becomes available at the next startup Upon system start up the Multiple Interface table is parsed and passes comprehensive validation tests If any errors occur during this validation phase the device sends an error message to the Syslog server and falls back to a safe mode using a single interface and no VLANs Please be sure to follow the Syslog messages that the device sends in system startup to see if any errors occurred When configuring the device using the Web interface it is possible to perform a quick validation of the configured Multiple Interface table and VLAN definitions by clicking the Done button in the Multiple Interface Table Web page It is highly recommended to perform this when configuring Multiple Interfaces and VLANs using the Web Interface to ensure the configuration is complete and valid 10 8 1 5 Troubleshooting the Multiple Interface Table If any of the Multiple Interface table guidelines are violated the device falls back to a safe mode configuration consisting of a single IPv4 interface and no VLANs For more information on validation failures consult the Syslog messages Validation failures may be caused by one of the following m One of the Application Types OAMP CON
545. ress gt m SASProxySet 1 Version 6 0 383 March 2010 9 3 7a e AudioCodes MediaPack Series Configuring DTMF Transport Types You can control the way DTMF digits are transported over the IP network to the remote endpoint by using one of the following modes Using INFO message according to Nortel IETF draft DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 e TxDTMFOption 1 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 Using INFO message according to Cisco s mode DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 e TxDTMFOption 3 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 Using NOTIFY messages according to lt draft mahy sipping signaled digits 01 txt gt DTMF digits are carried to the remote side using NOTIFY messages To enable this mode define the following e RxDTMFOption 0 e TxDTMFOption 2 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 Using RFC 2833 relay with Payload type negotiation DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard To enable this mode define the following
546. resses and UDP ports You can disable the NAT mechanism by setting the ini file parameter DisableNAT to 1 The two parameters EnablelpAddrTranslation and EnableUdpPortTranslation allow you to specify the type of compare operation that occurs on the first incoming packet To compare only the IP address set EnablelpAddrTranslation to 1 and EnableUdpPortTranslation to 0 In this case if the first incoming packet arrives with only a difference in the UDP port the sending addresses won t change If both the IP address and UDP port need to be compared then both parameters need to be set to 1 Version 6 0 445 March 2010 7a L tal AudioCodes MediaPack Series 10 2 3 No Op Packets 10 3 10 4 The device s No Op packet support can be used to verify Real Time Transport Protocol RTP and T 38 connectivity and to keep NAT bindings and Firewall pinholes open The No Op packets are available for sending in RTP and T 38 formats You can control the activation of No Op packets by using the ini file parameter NoOpEnable If No Op packet transmission is activated you can control the time interval in which No Op packets are sent in the case of silence i e no RTP or T 38 traffic This is performed using the ini file parameter NoOplnterval For a description of the RTP No Op ini file parameters refer to Networking Parameters on page 207 m RTP No Op The RTP No Op support complies with IETF s draft wing avt rtp noop 03 txt titled A No O
547. retransmissions of SIP messages The default is 4000 Note The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx Maximum number of UDP transmissions first transmission plus retransmissions of SIP messages The range is 1 to 30 The default value is 7 Number of retransmitted INVITE REGISTER messages before the call is routed hot swap to another Proxy Registrar The valid range is 1 to 30 The default value is 3 Note This parameter is also used for alternative routing using the Tel to IP Routing If a domain name in the table is resolved into two IP addresses and if there is no response for HotSwapRtx retransmissions to the INVITE message that is sent to the first IP address the device immediately initiates a call to the second IP address IP Group Proxy Registration and Authentication Parameters The proxy server registration and authentication SIP parameters are described in the table below Table 6 28 Proxy Registration and Authentication SIP Parameters Parameter Web IP Group Table EMS Endpoints gt IP Group IPGroup Version 6 0 Description This ini file table parameter configures the IP Group table The format of this parameter is as follows IPGroup FORMAT IPGroup Index IPGroup Type IPGroup Description IPGroup ProxySetld IPGroup SIPGroupName IPGroup ContactUser IPGroup EnableSurvivability IPG
548. rings default 1 Not ring related Note For this parameter to take effect a device reset is required 300 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 8 5 Call Hold Parameters The call hold parameters are described in the table below Parameter Web EMS Enable Hold EnableHold Web EMS Hold Format HoldFormat Web EMS Held Timeout HeldTimeout Web Call Hold Reminder Ring Timeout EMS CHRR Timeout CHRRTimeout Version 6 0 Table 6 38 Call Hold Parameters Description Allows users connected to the device to place a call on hold 0 Disable Disables the Hold service 1 Enable Enables the Hold service default If the Hold service is enabled a user can place the call on hold or remove from hold using the Hook Flash button On receiving a Hold request the remote party is placed on hold and hears the hold tone Note To use this service the devices at both ends must support this option Determines the format of the SDP in the Re INVITE hold request 0 0 0 0 0 The SDP c field contains the IP address 0 0 0 0 and the a inactive attribute default 1 Send Only The SDP c field contains the device s IP address and the a sendonly attribute Note The device does not send any RTP packets when it is in hold state for both hold formats Determines the time interval that the device can allow a call to remain on h
549. rio appears with its Steps in the Navigation tree 3 Click the Edit Scenario button located at the bottom of the Navigation pane the Scenario Name and Step Name fields appear 4 You can perform the following edit operations e Add Steps a On the Navigation bar select the desired tab i e Configuration or Management the tab s menu appears in the Navigation tree b Inthe Navigation tree navigate to the desired page item the corresponding page opens in the Work pane c Inthe page select the required parameter s by marking the corresponding check box es d Click Next o Add or Remove Parameters a Inthe Navigation tree select the required Step the corresponding page opens in the Work pane b To add parameters select the check boxes corresponding to the desired parameters to remove parameters clear the check boxes corresponding to the parameters that you want removed c Click Next SIP User s Manual 40 Document LTRT 65413 SIP User s Manual 3 Web Based Management e Edit the Step Name a Inthe Navigation tree select the required Step b Inthe Step Name field modify the Step name c Inthe page click Next e Edit the Scenario Name a Inthe Scenario Name field edit the Scenario name b Inthe displayed page click Next e Remove a Step a Inthe Navigation tree select the required Step the corresponding page opens in the Work pane b In the page clear all the check box
550. rname Register ContactUser Application Type 1 Oj Version 6 0 93 March 2010 Ao c tal AudioCodes MediaPack Series 2 To add an Account in the Add field enter the desired table row index and then click Add A new row appears 3 Configure the Account parameters according to the table below 4 Click the Apply button to save your changes 5 To save the changes refer to Saving Configuration on page 161 Note For a description of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 34 Table 3 15 Account Table Parameters Description Parameter Description Served Trunk Group The Hunt Group ID for which the device performs registration Account_ServedTrunkGroup and or authentication to a destination IP Group i e Serving IP Group For Tel to IP calls the Served Hunt Group is the source Hunt Group from where the call initiated For IP to Tel calls the Served Hunt Group is the Hunt Group ID defined in the IP to Hunt Group Routing Table refer to Configuring the IP to Hunt Group Routing Table on page 131 For defining Hunt Groups refer to Configuring Endpoint Phone Numbers on page 143 Serving IP Group The destination IP Group ID defined in Configuring the IP Account ServinglPGroup Groups on page 91 to where the REGISTER reguests if enabled are sent or Authentication is performed The actual destination to where the REGISTER re
551. round than the advanced parameters Version 6 0 31 March 2010 7a K tal AudioCodes MediaPack Series When the Navigation tree is in Full mode refer to Navigation Tree on page 27 configuration pages display all their parameters i e the Advanced Parameter List view is displayed If a page contains only basic parameters the Basic Parameter List button is not displayed 3 1 6 2 2 Showing Hiding Parameter Groups Some pages provide groups of parameters which can be hidden or shown To toggle between hiding and showing a group simply click the group name button that appears above each group The button appears with a down pointing or up pointing arrow indicating that it can be collapsed or expanded when clicked respectively Figure 3 8 Expanding and Collapsing Parameter Groups 3 1 6 3 Modifying and Saving Parameters When you change parameter values on a page the Edit symbol appears to the right of these parameters This is especially useful for indicating the parameters that you have currently modified before applying the changes After you save your parameter modifications refer to the procedure described below the Edit symbols disappear Figure 3 9 Editing Symbol after Modifying Parameter Value prance Setongs N Basic Parameter List a v General Settings A Dynamic Jitter Buffer Minimum Delay 8 Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Edit Symbol Packing
552. roup ServinglPGroup IPGroup SipReRoutingMode IPGroup AlwaysUseRouteTable IPGroup RoutingMode IPGroup SRD IPGroup MediaRealm IPGroup ClassifyByProxySet IPGroup Profileld MPGroup For example IPGroup 1 0 dol gateway 1 firstIPgroup 0 1 0 0 1 0 1 1 IPGroup 2 0 abc server 2 secondlPgroup 0 1 0 0 1 0 1 2 IPGroup 3 1 IP phones 1 thirdiPGroup 0 1 0 0 1 0 mere 263 March 2010 A K tal AudioCodes MediaPack Series Parameter Description Notes This table parameter can include up to 9 indices 1 9 The parameters Type EnableSurvivability ServinglPGroup RoutingMode SRD MediaRealm and ClassifyByProxySet are not applicable For a detailed description of the ini file table s parameters and for configuring this table using the Web interface refer to Configuring the IP Groups on page 91 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Authentication Table EMS SIP Endpoints gt Authentication Authentication This ini file table parameter defines a user name and password for authenticating each device port The format of this parameter is as follows Authentication FORMAT Authentication_Index Authentication_Userld Authentication UserPassword Authentication Where Index port number where O0 depicts the Port 1 Userld User name UserPassword Password
553. rport value of the response to the actual port from where the request was received This method is used for example to enable the device to identify its port mapping outside a NAT If the Via header doesn t include the rport parameter the destination port of the response is obtained from the host part of the Via header If the Via header includes the rpor parameter without a port value the destination port of the response is the source port of the incoming request If the Via header includes rport with a port value e g rport 1001 the destination port of the response is the port indicated in the rpor parmeter Determines whether the SIP X Channel header is added to SIP messages for providing information on the physical channel on which the call is received or placed 0 Disable X Channel header is not used default 1 Enable X Channel header is generated by the device and sent in INVITE messages and 180 183 and 200 OK SIP responses The header includes the channel and the device s IP address For example x channel DS DS1 1 8 IP 192 168 13 1 where DS DS 1 is a constant string 1 is a constant string 8 is the channel port IP 192 168 13 1 is the device s IP address LANA 297 March 2010 ca AudioCodes Parameter Web EMS Progress Indicator to IP ProgressIndicator2IP EnableRekeyAfter181 NumberOfActiveDialogs Web EMS Default Release Cause DefaultReleaseCause
554. rs associated with dual tone multi freguency DTMF and dialing For a description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt To configure the DTMF and dialing parameters 1 Open the DTMF 8 Dialing page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt DTMF 8 Dialing page item Figure 3 59 DTMF amp Dialing Page vw Inter Digit Timeout sec 4 Declare RFC 2833 in SDP Yes 1st Tx DTMF Option RFC 2833 2nd Tx DTMF Option RFC 2833 Payload Type 101 Hook Flash Option Not Supported Digit Mapping Rules Dial Plan Index 1 Dial Tone Duration sec 16 Hotline Dial Tone Duration sec 16 Enable Special Digits Disable Default Destination Number 1000 Special Digit Representation Special Max Digits In Phone Num 30 2 3 4 Configure the parameters as required Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 4 Proxies Registration IP Groups The Proxies Registration IP Groups submenu allows you to configure SIP proxy servers registration parameters and IP Groups This submenu includes the following items m IP Group Table refer to Configuring the IP Groups on page 91 m Account Table refer to Configuring the Account Table on
555. rs notifying you of this Throughout the Web interface parameters preceded by the lightning symbol are not applied on the fly and require that you reset the device for them to take effect When you modify parameters that require a device reset once you click the Submit button in the relevant page the toolbar displays the word Reset refer to Toolbar on page 26 to indicate that a device reset is required SIP User s Manual 160 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 4 1 3 2 Locking and Unlocking the Device The Lock and Unlock options allow you to lock the device so that it doesn t accept any new incoming calls This is useful when for example you are uploading new software files to the device and you don t want any traffic to interfere with the process gt To lock the device 1 Open the Maintenance Actions page refer to Maintenance Actions on page 159 2 Under the LOCK UNLOCK group from the Graceful Option drop down list select one of the following options e Yes The device is locked only after the user defined time in the Lock Timeout field refer to Step 3 expires or no more active traffic exists the earliest thereof In addition no new traffic is accepted e No The device is locked regardless of traffic Any existing traffic is terminated immediately Note These options are only available if the current status of the device is in the Unlock st
556. rst Ringback Tone in the CPT file This option enables an Application server to request the device to play a distinctive Ringback tone to the calling party according to the destination of the call The tone is played according to the Alert Info header received in the 180 Ringing SIP response the value of the Alert Info header is added to the value of this parameter The valid range is 1 to 1 000 The default value is 1 i e play standard Ringback tone Notes tis assumed that all Ringback tones are defined in sequence in the CPT file Incase of an MLPP call the device uses the value of this parameter plus 1 as the index of the Ringback tone in the CPT file e g if this value is set to 1 then the index is 2 i e 1 1 The time interval from when the user hangs up the phone until the call is disconnected FXS This allows the user to hang up and then pick up the phone before this timeout to continue the call conversation Thus it s also referred to as regret time The valid range is 0 to 255 in seconds The default value is 0 259 March 2010 ca AudioCodes Parameter Web Enable Reanswering Info EnableReansweringINFO Web EMS SIT Q850 Cause SITQ850Cause Web EMS SIT Q850 Cause For NC SITQ850CauseForNC SIP User s Manual MediaPack Series Description Enables the device to send a SIP INFO message with the On Hook Off Hook parameter when the FXS phone goes on hook during an ongoing call an
557. rt on page 408 SIP User s Manual 62 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 3 2 3 Configuring the RTP RTCP Settings The RTP RTCP Settings page allows you to configure the Real Time Transport Protocol RTP and Real Time Transport RTP Control Protocol RTCP parameters For a detailed description of the parameters appearing on this page refer to Configuration Parameters Reference on page 207 gt To configure the RTP RTCP parameters 1 Open the RTP RTCP Settings page Configuration tab gt Media Settings menu gt RTP RTCP Settings page item Figure 3 42 RTP RTCP Settings Page General Settings Dynamic Jitter Buffer Minimum Delay Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Packing Factor Basic RTP Packet Interval RFC 2833 TX Payload Type 96 RFC 2833 RX Payload Type 96 RFC 2198 Payload Type 104 Fax Bypass Payload Type 102 Enable RFC 3389 CN Payload Type Enable RTP Base UDP Port 6000 Comfort Noise Generation Negotiation Disable Analog Signal Transport Type Disable Remote RTP Base UDP Port RTP Multiplexing Local UDP Port RTP Multiplexing Remote UDP Port 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on
558. rt beeps and for Bellcore and ETSI Caller IDs can view the Caller ID string of the incoming call The calling party hears a Call Waiting Ringback Tone The called party can accept the new call using hook flash and can toggle between the two calls gt To enable call waiting 1 Set the parameter EnableCallWaiting to 1 2 Set the parameter EnableHold to 1 3 Define the Call Waiting indication and Call Waiting Ringback tones in the Call Progress Tones file You can define up to four Call Waiting indication tones refer to the parameter FirstCallWaitingTonelD in SIP Configuration Parameters on page 245 4 To configure the Call Waiting indication tone cadence modify the following parameters NumberOfWaitingIndications WaitingBeepDuration and TimeBetweenWaitingIndications Version 6 0 415 March 2010 7a e AudioCodes MediaPack Series 9 7 7 9 7 8 9 7 8 1 5 To configure a delay interval before a Call Waiting Indication is played to the currently busy port use the parameter TimeBeforeWaitinglndication This enables the caller to hang up before disturbing the called party with Call Waiting Indications Applicable only to FXS modules Both the calling and called sides are supported by FXS interfaces FXO interfaces support only the calling side To indicate Call Waiting the device sends a 182 Call Queued response The device identifies Call Waiting when a 182 Call Queued response is received Message Waiting In
559. rver 1P Address Suse Gor s Manogement Settings Syslog Server Port Management Configuration Enable Syslog Regione Setbogs i Maemtenance Acbons software Update A Activity Types to Report vie Activity Log Messages The Web GUI is composed of the following main areas m Title bar Displays the corporate logo and product name m Toolbar Provides frequently required command buttons for configuration refer to Toolbar on page 26 m Navigation Pane Consists of the following areas e Navigation bar Provides tabs for accessing the configuration menus refer to Navigation Tree on page 27 creating a Scenario refer to Scenarios on page 37 and searching ini file parameters that have corresponding Web interface parameters refer to Searching for Configuration Parameters on page 35 e Navigation tree Displays the elements pertaining to the tab selected on the Navigation bar tree like structure of the configuration menus Scenario Steps or Search engine m Work pane Displays configuration pages where all configuration is performed refer to Working with Configuration Pages on page 29 Version 6 0 25 March 2010 7a K tal AudioCodes MediaPack Series 3 1 4 Toolbar The toolbar provides command buttons for quick and easy access to frequently required commands as described in the table below Table 3 1 Description of Toolbar Buttons Icon Button Description Name y Submit Applies parameter sett
560. ry files are loaded to the device Loading a Software Upgrade Key The Software Upgrade Key Status page allows you to load a new Software Upgrade Key to the device The device is supplied with a Software Upgrade Key which determines the device s supported features capabilities and available resources You can upgrade or change your device s supported items by purchasing a new Software Upgrade Key to match your requirements The Software Upgrade Key is provided in string format in a text based file When you load a Software Upgrade Key it is loaded to the device s non volatile flash memory and overwrites the previously installed key You can load a Software Upgrade Key using one of the following management tools m Web interface m BootP TFTP configuration utility refer to Loading via BootP TFTP on page 167 m AudioCodes EMS refer to EMS User s Manual or EMS Product Description Warning Do not modify the contents of the Software Upgrade Key file Note The Software Upgrade Key is an encrypted key The procedure below describes how to load a Software Upgrade Key to the device using the Web interface 165 March 2010 7a c tall AudioCodes MediaPack Series gt To load a Software Upgrade Key 1 Open the Software Upgrade Key Status page Management tab gt Software Update menu gt Software Upgrade Key page item Figure 3 105 Software Upgrade Key Page Current Key Key features Board Type
561. s Physically disconnected from the network i e Ethernet cable is disconnected 261 March 2010 ca AudioCodes Parameter Web Out Of Service Behavior EMS FXS OOS Behavior FXSOOSBehavior Retransmission Parameters Web SIP T1 Retransmission Timer msec EMS T1 RTX SipT1Rtx SIP User s Manual MediaPack Series Description The Ethernet cable is connected but the device can t communicate with any host Note that LAN Watch Dog must be activated the parameter EnableLANWatchDog set to 1 The device can t communicate with the proxy according to the Proxy Keep Alive mechanism and no other alternative route exists to send the call The IP Connectivity mechanism is enabled using the parameter AltRoutingTel2IPEnable and there is no connectivity to any destination IP address Notes The FXSOOSBehavior parameter determines the behavior of the FXS endpoints when a Busy Out or Graceful Lock occurs FXO endpoints during Busy Out and Lock are inactive Refer to the LifeLineType parameter for complementary optional behavior Determines the behavior of undefined FXS endpoints and all FXS endpoints when a Busy Out condition exists 0 None Normal operation No response is provided to undefined endpoints A dial tone is played to FXS endpoints when a Busy Out condition exists 1 Reorder Tone The device plays a reorder tone to the connected phone PBX default 2 Polarity Reversal
562. s 4 To save the changes to flash memory refer to Saving Configuration on page 161 Version 6 0 65 March 2010 7a tal AudioCodes MediaPack Series 3 3 3 3 3 3 1 Access Level Security Administrator Security Settings The Security Settings menu allows you to configure various security settings This menu contains the following page items m Web User Accounts refer to Configuring the Web User Accounts on page 66 m WEB amp Telnet Access List refer to Configuring the Web and Telnet Access List on page 69 Firewall Settings refer to Configuring the Firewall Settings on page 70 Certificates refer to Configuring the Certificates on page 73 802 1x Settings refer to Configuring the 802 1x Settings on page 77 General Security Settings refer to Configuring the General Security Settings on page 78 m PSec Proposal Table refer to Configuring the IP Security Associations Table on page 80 m IPSec Association Table refer to Configuring the IP Security Proposal Table on page 79 Configuring the Web User Accounts To prevent unauthorized access to the Web interface two Web user accounts are available primary and secondary with assigned user name password and access level When you login to the Web interface you are requested to provide the user name and password of one of these Web user accounts If the Web session is idle i e no actions are performed for more tha
563. s The Maintenance Actions page allows you to perform the following operations Reset the device refer to Resetting the Device on page 159 Lock and unlock the device refer to Locking and Unlocking the Device on page 161 Save the configuration to the device s flash memory refer to Saving Configuration on page 161 To access the Maintenance Actions page On the Navigation bar click the Management tab and then in the Navigation tree select the Management Configuration menu and then choose the Maintenance Actions page item Figure 3 101 Maintenance Actions Page w Reset Configuration Burn To FLASH Yes Graceful Option No LOCK UNLOCK Lock LOCK Graceful Option No Current Adrnin State UNLOCKED w Save Configuration 3 4 1 3 1 Resetting the Device The Maintenance Actions page allows you to remotely reset the device In addition before resetting the device you can choose the following options Version 6 0 Save the device s current configuration to the device s flash memory non volatile Perform a graceful shutdown i e device reset starts only after a user defined time expires i e timeout or after no more active traffic exists the earliest thereof To reset the device Open the Maintenance Actions page refer to Maintenance Actions on page 159 Under the Reset Configuration group from the Burn To FLASH drop down list select one of th
564. s The signal tone detection parameters are described in the table below Table 6 49 Tone Detection Parameters Parameter EMS DTMF Enable DTMF DetectorEnable EMS MF R1 Enable MFR1DetectorEnable EMS User Defined Tone Enable UserDefinedToneDetectorEnable EMS SIT Enable SITDetectorEnable Version 6 0 Description Enables or disables the detection of DTMF signaling 0 Disable 1 Enable default Enables or disables the detection of MF R1 signaling 0 Disable default 1 Enable Enables or disables the detection of User Defined Tones signaling applicable for Special Information Tone SIT detection 0 Disable default 1 Enable Enables or disables SIT detection according to the ITU T recommendation E 180 Q 35 0 Disable default 1 Enable applicable to FXO interfaces SlTDetectorEnable 1 UserDefinedToneDetectorEnable 1 DisconnectOnBusyTone 1 applicable for Busy Reorder and SIT tones Note For this parameter to take effect a device reset is required 321 March 2010 A c tal AudioCodes MediaPack Series Parameter Description EMS UDT Detector Frequency Defines the deviation in Hz allowed for the detection of each Deviation signal frequency UDTDetectorFrequencyDeviation The valid range is 1 to 50 The default value is 50 Note For this parameter to take effect a device reset is required EMS CPT Detect
565. s refer to Dialing Plan Notation for Routing and Manipulation on page 377 All calls matching all or any combination of the above characteristics are sent to the destination IP address defined below Note For alternative routing additional entries of the same prefix can be configured Web Dest IP Address EMS Address Version 6 0 Destination IP address in dotted decimal notation or FQDN to where the call must be sent If an FQDN is used e g domain com DNS resolution is performed according to the parameter DNSQueryType Notes f you defined a destination IP Group above then this IP address is not used for routing and therefore not required To discard these calls enter 0 0 0 0 For example if you want to prohibit dialing of International calls then in the Dest Phone Prefix field enter 00 and in the Dest IP Address field enter 0 0 0 0 For routing calls between phones connected to the device i e local routing enter the device s IP address When the device s IP address is unknown e g when DHCP is used enter IP address 127 0 0 1 When using domain names you must enter the DNS server s IP address or alternatively define these names in the Internal DNS Table refer to Configuring the Internal DNS Table on page 134 Ifthe string ENUM is specified for the destination IP address an ENUM query containing the destination phone number is sent to the DNS server 129 March 201
566. s Ethernet connection can be configured using the ini file parameter EthernetPhyConfiguration for one of the following modes m Manual mode e 10Base T Full Duplex e 100Base TX Half Duplex or 100Base TX Full Duplex m Auto Negotiation chooses common transmission parameters such as speed and duplex mode The Ethernet connection should be configured according to the following recommended guidelines m When the device s Ethernet port is configured for Auto Negotiation the opposite port must also operate in Auto Negotiation Auto Negotiation falls back to Half Duplex mode when the opposite port is not in Auto Negotiation mode but the speed i e 10 100Base T or 1000Base TX in this mode is always configured correctly Configuring the device to Auto Negotiation mode while the opposite port is set manually to Full Duplex is invalid as it causes the device to fall back to Half Duplex mode while the opposite port is Full Duplex Any mismatch configuration can yield unexpected functioning of the Ethernet connection m When configuring the device s Ethernet port manually the same mode i e Half Duplex or Full Duplex and speed must be configured on the remote Ethernet port In addition when the device s Ethernet port is configured manually it is invalid to set the remote port to Auto Negotiation Any mismatch configuration can yield unexpected functioning of the Ethernet connection m It s recommended to configure the port for best performan
567. s Manual 3 Web Based Management Parameter Description Trap Port Defines the port number of the remote SNMP Manager SNMPManagerTrapPort_x The device sends SNMP traps to these ports The valid SNMP trap port range is 100 to 4000 The default port is 162 Trap Enable Activates or de activates the sending of traps to the SNMPManagerTrapSendingEnable_x corresponding SNMP Manager 0 Disable Sending is disabled 1 Enable Sending is enabled default 3 4 1 1 2 Configuring the SNMP Community Strings Version 6 0 The SNMP Community String page allows you to configure up to five read only and up to five read write SNMP community strings and to configure the community string that is used for sending traps For detailed information on SNMP community strings refer to the Product Reference Manual gt 1 To configure the SNMP community strings Access the Management Settings page as described in Configuring the Management Settings on page 152 In the SNMP Community String field click the right pointing arrow 2 button the SNMP Community String page appears Figure 3 97 SNMP Community Strings Page Delete Community String Access Level Read Only Read Only Read Only Read Only Read Only Read Write Read Write Read Write Read Write Trap Community String trapuser i Read Write DDDO ODO DN
568. s NLP Enables or disables the Aggressive NLP at the first 0 5 second of the call When enabled the echo is removed only in the first half of a second of the incoming IP signal 0 Disable 1 Enable default Note For this parameter to take effect a device reset is required Determines whether Silence Indicator SID packets are sent according to RFC 3389 0 Disable G 711 SID packets are sent in a proprietary method default 1 Enable SID comfort noise packets are sent with the RTP SID payload type according to RFC 3389 This is applicable only to G 711 and G 726 coders Determines the number of spectral coefficients added to an SID packet being sent according to RFC 3389 Valid only if EnableStandardSIDPayloadType is set to 1 The valid values are 0 default 4 6 8 and 10 350 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 16 2 Fax and Modem Parameters The fax and modem parameters are described in the table below Table 6 58 Fax and Modem Parameters Parameter Web Fax Transport Mode EMS Transport Mode FaxTransportMode Web Fax Relay Enhanced Redundancy Depth EMS Enhanced Relay Redundancy Depth FaxRelayEnhancedRedundancy Depth Web Fax Relay Redundancy Depth EMS Relay Redundancy Depth FaxRelayRedundancyDepth Web Fax Relay Max Rate bps EMS Relay Max Rate FaxRelayMaxRate Version 6 0 Description Fax trans
569. s Reference on page 207 gt To configure the fax modem and CID parameters 1 Open the Fax Modem CID Settings page Configuration tab gt Media Settings menu gt Fax Modem CID Settings page item Figure 3 41 Fax Modem CID Settings Page w General Settings Fax Transport Mode RelayEnable Caller ID Transport Type Mute Caller ID Type Standard Bellcore V 21 Modem Transport Type Disable v 22 Modem Transport Type Enable Bypass 23 Modem Transport Type Enable Bypass V 32 Modem Transport Type Enable Bypass V 34 Modem Transport Type Enable Bypass Fax CNG Mode Disable SNS S SNS NS SNES NS CNG Detector Mode Disable w Fax Relay Settings Fax Relay Redundancy Depth 0 Fax Relay Enhanced Redundancy Depth 4 Fax Relay ECM Enable Enable Fax Relay Max Rate bps 14400bps T38 Version T 38 version 0 v Bypass Settings Fax Modem Bypass Coder Type G 11Alaw_64 Fax Modem Bypass Packing Factor Fax Bypass Output Gain Modem Bypass Output Gain 2 Configure the parameters as required 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Note Some SIP parameters override these fax and modem parameters refer to the parameter IsFaxUsed and V 152 parameters in Section V 152 Suppo
570. s a prefix that is added to the Reguest URI user part of the INVITE message that is sent by the device s SAS agent when in Emergency mode to the default gateway or to any other destination using the IP2IP Routing table This parameter is reguired to differentiate between normal SAS calls routed to the default gateway and emergency SAS calls Therefore this allows you to define different manipulation rules for normal and emergency calls This valid value is a character string The default is an empty string Web SAS Registration Manipulation Table EMS Stand Alone Survivability SASRegistrationManipulation SIP User s Manual This ini file table parameter configures the SAS Registration Manipulation table This table is used by the SAS application to manipulate the user part of an incoming REGISTER request AoR the To header before saving it to the registered users database The format of this table parameter is as follows SASRegistrationManipulation FORMAT SASRegistrationManipulation Index SASRegistrationManipulation RemoveFromRight SASRegistrationManipulation LeaveFromRight SASRegistrationManipulation RemoveFromRight number of digits removed from the right side of the user part before saving to the registered user database 312 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web SAS IP to IP Routing Table IP2IPRouting Version 6 0 Description
571. s changed to E164 National gt To configure the Phone Context tables 1 Open the Phone Context Table page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt Phone Context page item Figure 3 76 Phone Context Table Page v Add Phone Context As Prefix Enable Phone Context Index 1 10 NPI TON i Phone Context Unknown Unknown unknown com Private Level 2 Regional host com E 164 Public National na e164 host com 2 Configure the Phone Context table according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Several rows with the same NPI TON or Phone Context are allowed In such a scenario a Tel to IP call uses the first match Phone Context is a unique case as it doesn t appear in the Request URI as a Phone Context parameter Instead it s added as a prefix to the phone number The isn t removed from the phone number in the IP to Tel direction You can also configure the Phone Context table using the ini file table parameter PhoneContext refer to Number Manipulation and Routing Parameters on page 331 SIP User s Manual 122 Document LTRT 65413 SIP User s Manual 3 Web Based Management Table 3 20 Phone Context Parameters Description Parameter Add Phone Context As Prefi
572. s parameter to take effect a device reset is required Determines the value of the RS 232 flow control 0 None default 1 Hardware Note For this parameter to take effect a device reset is required BootP Parameters The BootP parameters are described in the table below The BootP parameters are special hidden parameters Once defined and saved in the device s flash memory they are used even if they don t appear in the ini file Parameter BootPRetries SIP User s Manual Table 6 17 BootP Parameters Description Note For this parameter to take effect a device reset is required This parameter is used to Sets the number of BootP requests the device sends during start up The device stops sending BootP requests when either BootP reply is received or number of retries is reached 1 1 BootP retry 1 sec 2 2 BoofP retries 3 sec 3 3 Boo P retries 6 sec default 4 10 BootP retries 30 sec 5 20 BootP retries 60 sec 6 40 BootP retries 120 sec 7 100 BootP retries 300 sec 230 Sets the number of DHCP packets the device sends If after all packets are sent there s still no reply the device loads from flash 1 4 DHCP packets 2 5 DHCP packets 3 6 DHCP packets default 4 7 DHCP packets 5 8 DHCP packets 6 9 DHCP packets 7 10 DHCP packets 15 18 DHCP packets Document LTRT 6
573. s the time interval in seconds that the NTP client requests EMS Update Interval for a time update NTPUpdatelnterval The default interval is 86400 i e 24 hours The range is 0 to 214783647 Note It is not recommend to set this parameter to beyond one month i e 2592000 seconds Daylight Saving Time Parameters Web Day Light Saving Time Determines whether to enable daylight saving time EMS Mode DayLightSavingTimeEnable DI Disable xdetauly 1 Enable Web Start Time Defines the date and time when daylight saving begins EMS Start The format of the value is mo dd hh mm month day hour and DayLightSavingTimeStart minutes Web End Time Defines the date and time when daylight saving ends EMS End The format of the value is mo dd hh mm month day hour and DayLightSavingTimeEnd minutes Web EMS Offset Daylight saving time offset in minutes DayLightSavingTimeOffset The valid range is 0 to 120 The default is 60 Version 6 0 221 March 2010 ca AudioCodes MediaPack Series 6 2 Web and Telnet Parameters This subsection describes the device s Web and Telnet parameters 6 2 1 General Parameters The general Web and Telnet parameters are described in the table below Parameter Web Web and Telnet Access List Table EMS Web Access Addresses WebAccessList x Web Use RADIUS for Web Telnet Login EMS Web Use Radius Login WebRADIUSLogin SIP User s Manual Table 6 10 General Web an
574. sDiffServ 10 Application Type for applications EnableDNSasOAM 1 EnableNTPasOAM 1 Multiple Interface Table Configuration InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes InterfaceTable InterfaceMode InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway InterfaceTable VlanID InterfaceTable InterfaceName mteertacerable 6 10 192 168 85 14 i16 1924158940511 A iM This ini file shows the following m A Multiple Interface table with a single interface 192 168 85 14 16 OAMP Media and Control applications are allowed and a default gateway 192 168 0 1 m A Routing table is configured with two routing rules directing all traffic for subnet 201 201 0 0 16 to 192 168 0 2 and all traffic for subnet 202 202 0 0 16 to 192 168 0 3 m VLANs are disabled Native VLAN ID is set to 1 m Values for the Class Of Service parameters are assigned m The DNS application is configured to act as an OAMP application and the NTP application is configured to act as an OAMP application SIP User s Manual 460 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities Lines that begin with a semicolon are considered a remark and are ignored The Multiple Interface table configuration using the ini file must have the prefix and suffix to allow AudioCodes INI File parser to correctly recognize the Multiple Interface Table The following sections sho
575. sable Disables DID Wink default 1 Enable Enables DID Wink If enabled the device can be used for connection to EIA TIA 464B DID Loop Start lines Both FXO detection and FXS generation are supported An FXO interface dials DTMF digits after a Wink signal is detected instead of a Dial tone An FXS interface generates the Wink signal after the detection of off hook instead of playing a Dial tone Defines the time interval in msec between detection of off hook and generation of a DID Wink The valid range is 0 to 1 000 The default value is 0 Note This parameters is applicable only to FXS interfaces Determines the type of DID signaling support for NTT Japan modem DTMF or Frequency Shift Keying FSK based signaling The devices can be connected to Japan s NTT PBX using Modem DID lines These DID lines are used to deliver a called number to the PBX 0 FSK based signaling default 1 DTMF based signaling Note This parameter is applicable only to FXS interfaces This ini file table parameter enables support for Japan NTT Modem DID FXS interfaces can be connected to Japan s NTT PBX using Modem DID lines These DID lines are used to deliver a called number to the PBX The DID signal can be sent alone or combined with an NTT Caller ID signal The format of this parameter is as follows EnableDID FORMAT EnableDID Index EnableDID IsEnable EnableDID Where Index Port number wh
576. sage that appears after a successful login to the Web interface The format of this parameter is as follows WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_ Text WelcomeMessage For Example WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_ Text WelcomeMessage 1 nkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkii WelcomeMessage 2 This is a Welcome message WelcomeMessage Note Each index represents a line of text in the Welcome message box Up to 20 indices can be defined SIP User s Manual 44 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 10 Getting Help The Web interface provides you with context sensitive Online Help The Online Help provides you with brief descriptions of most of the parameters you ll need to successfully configure the device The Online Help provides descriptions of parameters pertaining to the currently opened page gt 1 To view the Help topic for a currently opened page Using the Navigation tree open the required page for which you want Help f i im On the toolbar click the Help button the Help topic pertaining to the opened page appears as shown below Figure 3 23 Help Topic for Current Page Help v NTP Settings NTP Server IP Address NTP UTC Offset NTP Updated interval v Day Light Saving Time Day Ught Saving Tene Start Tene End Tene Offset min Help Topics To view a description
577. save the configuration to the device s flash memory This enables the device to use a temporary IP address for initial management and configuration while retaining the address defined in this table for deployment For an explanation on configuring tables in the Web interface refer to Working with Tables on page 34 You can also configure this table using the ini file table parameter InterfaceTable refer to Networking Parameters on page 207 gt To configure the multiple IP interface table 1 Open the IP Settings page Configuration tab gt Network Settings menu gt IP Settings Figure 3 33 IP Settings Page Single IP Settings IP Address 10 8 6 31 Subnet Mask 255 255 0 0 Default Gateway Address 10 8 0 1 i v Multiple Interface Settings Multiple Interface Table 2 Under the Multiple Interface Settings group click the Multiple Interface Table um button a confirmation message box appears Figure 3 34 Confirmation Message for Accessing the Multiple Interface Table Microsoft Internet Explorer 2 If switching to the advanced interface configuration mode the current page wil no longer be available Are you sure you want to continue Cra Version 6 0 51 March 2010 Aa c tal AudioCodes MediaPack Series 3 Click OK to confirm the Multiple Interface Table page appears Figure 3 35 Multiple Interface Table Page Index Application Type IP Address Gate
578. se SIP Tgrp UseSIPTgrp SIP User s Manual MediaPack Series Description Enables usage of the History Info header 0 Disable default 1 Enable User Agent Client UAC Behavior Initial request The History Info header is equal to the Request URI If a PSTN Redirect number is received it is added as an additional History Info header with an appropriate reason Upon receiving the final failure response the device copies the History Info as is adds the reason of the failure response to the last entry and concatenates a new destination to it if an additional request is sent The order of the reasons is as follows a Q 850 Reason b SIP Reason c SIP Response code Upon receiving the final response success or failure the device searches for a Redirect reason in the History Info i e 3xx 4xx SIP reason If found it is passed to ISDN according to the following table SIP Reason Code ISDN Redirecting Reason 302 Moved Temporarily Call Forward Universal CFU 408 Request Timeout Call Forward No Answer CFNA 480 Temporarily Unavailable 487 Request Terminated 486 Busy Here Call Forward Busy CFB 600 Busy Everywhere f history reason is a Q 850 reason it is translated to the SIP reason according to the SIP ISDN tables and then to ISDN Redirect reason according to the table above User Agent Server UAS Behavior The History Info header is sent only in the final response Upon
579. se Tel Profiles to the device s channels in the Endpoint Phone Number Table page thereby applying different behaviors to different channels i e ports Note You can also configure Tel Profiles using the ini file table parameter TelProfile refer to SIP Configuration Parameters on page 245 gt To configure Tel Profiles 1 Open the Tel Profile Settings page Configuration tab gt Protocol Configuration menu gt Coders And Profile Definitions submenu gt Tel Profile Settings page item Figure 3 66 Tel Profile Settings Page v Profile ID Profile Name w Profile Parameters Profile Preference Fax Signaling Method Dynamic Jitter Buffer Minimum Delay msec Dynamic Jitter Buffer Optimization Factor RTP IP DiffServ Signaling DiffServ Voice Volume 32 to 31 dB DTMF Volume 31 to 0 dB Input Gain 32 to 31 dB Enable Digit Delivery Enable Polarity Reversal Enable Current Disconnect MWI Analog Lamp MWI Display Dial Plan Index Echo Canceler Flash Hook Period Enable Early Media Progress Indicator to IP Disconnect Call on Detection of Busy Tone Enable Voice Mail Delay Time For Reorder Tone sec Enable 911 PSAP Enable 4GC EC NLP Mode Swap Tel To IP Phone Numbers 10 10 46 40 0 1 0 Disable Disable Disable Disable Disable lt Enab
580. sed x Web IP Address EMS Address SNMPManagerTablelP x Web Trap Port EMS Port SNMPManagerTrapPort x Web Trap Enable SNMPManagerTrapSendingEnable x SNMPManagerTrapUser x Web Trap Manager Host Name SNMPTrapManagerHostName Version 6 0 Determines the validity of the parameters IP address and port number of the corresponding SNMP Manager used to receive SNMP traps 0 Check box cleared Disabled default 1 Check box selected Enabled Defines the IP address of the remote host used as an SNMP Manager The device sends SNMP traps to this IP address Enter the IP address in dotted decimal notation e g 108 10 1 255 Defines the port number of the remote SNMP Manager The device sends SNMP traps to this port The valid SNMP trap port range is 100 to 4000 The default port is 162 Activates or de activates the sending of traps to the corresponding SNMP Manager 0 Disable Sending is disabled 1 Enable Sending is enabled default This parameter can be set to the name of any configured SNMPV3 user to associate with this trap destination This determines the trap format authentication level and encryption level By default the trap is associated with the SNMP trap community string Defines an FQDN of a remote host that is used as an SNMP manager The resolved IP address replaces the last entry in the Trap Manager table defined by the parameter SNMP ManagerTablelP x and t
581. select the Delete Row check box that corresponds to the routing rule entry and then click Delete Selected Entries Table 3 6 IP Routing Table Description Parameter Description Destination IP Address Specifies the IP address of the destination host RoutingTableDestinationsColumn network Destination Mask Specifies the subnet mask of the destination host RoutingTableDestinationMasksColumn network SIP User s Manual 58 Document LTRT 65413 SIP User s Manual 3 Web Based Management Parameter Description The address of the host network you want to reach is determined by an AND operation that is applied to the fields Destination IP Address and Destination Mask For example to reach the network 10 8 x x enter 10 8 0 0 in the field Destination IP Address and 255 255 0 0 in the field Destination Mask As a result of the AND operation the value of the last two octets in the field Destination IP Address is ignored To reach a specific host enter its IP address in the field Destination IP Address and 255 255 255 255 in the field Destination Mask Gateway IP Address RoutingTableGatewaysColumn Metric RoutingTableHopsCountColumn Interface RoutingTablelnterfacesColumn Version 6 0 The IP address of the router next hop to which the packets are sent if their destination matches the rules in the adjacent columns Note The Gateway address must be in the same subnet on which t
582. selected all the calling party parameters are set from this header If P Asserted Identity is selected the Privacy header is checked and if the Privacy is set to id the calling number is assumed restricted FROM Use the source number received in the From header Determines the SIP header used for obtaining the called number destination for IP to Tel calls 0 Request URI header default Obtains the destination number from the user part of the Request URI Document LTRT 65413 254 SIP User s Manual Parameter Web EMS Forking Handling Mode ForkingHandlingMode Web Forking Timeout ForkingTimeOut Web EMS Enable Reason Header EnableReasonHeader Web EMS Gateway Name SIPGatewayName ZeroSDPHandling Version 6 0 6 Configuration Parameters Reference Description 1 To header Obtains the destination number from the user part of the To header 2 P Called Party ID header Obtains the destination number from the P Called Party ID header Determines how the device handles the receipt of multiple SIP 18x responses when forking is used by a Proxy for Tel to IP calls 0 Parallel handling The device opens a voice stream toward the first 18x SIP response that includes an SDP and disregards any 18x response with an SDP received thereafter default 1 Sequential handling The device opens a voice stream toward the first 18x SIP response that includes an SDP and r
583. selected folder Version 6 0 171 March 2010 7a e AudioCodes MediaPack Series 3 5 3 5 1 3 5 1 1 To load or restore the ini file 1 To load the ini file to the device perform the following 2 Click the Browse button navigate to the folder in which the ini file is located select the file and then click Open the name and path of the file appear in the field beside the Browse button 3 Click the Load INI File button and then at the prompt click OK the device uploads the ini file and then resets from the cmp version stored on the flash memory Once complete the Enter Network Password dialog box appears requesting you to enter your user name and password Status amp Diagnostics Tab The Status amp Diagnostics tab on the Navigation bar displays menus in the Navigation tree related to device operating status and diagnostics These menus include the following m Status amp Diagnostics refer to Status amp Diagnostics on page 172 m Gateway Statistics refer to Gateway Statistics on page 177 Status amp Diagnostics The Status amp Diagnostics menu is used to view and monitor the device s channels Syslog messages hardware and software product information and to assess the device s statistics and IP connectivity information This menu includes the following page items Message Log refer to Viewing the Device s Syslog Messages on page 172 Ethernet Port Information refer to Viewing Et
584. sends a 200 OK in response to an INVITE only when it detects the start of speech or ringback tone from the Tel side Note that the IPM detectors must be enabled 9 4 2 1 2 Two Stage Dialing Two stage dialing is when the IP caller is required to dial twice The caller initially dials to the FXO device and only after receiving a dial tone from the PBX via the FXO device dials the destination telephone number Figure 9 5 Call Flow for Two Stage Dialing FXO gam amp Client F1 INVITE FXO seizes line Two stage dialing implements the Dialing Time feature Dialing Time allows you to define the time that each digit can be separately dialed By default the overall dialing time per digit is 200 msec The longer the telephone number the greater the dialing time The relevant parameters for configuring Dialing Time include the following m DTMFDigitLength 100 msec time for generating DTMF tones to the PSTN PBX side m DTMFinterDigitInterval 100 msec time between generated DTMF digits to PSTN PBX side SIP User s Manual 388 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 4 2 1 3 DID Wink The device s FXO ports support Direct Inward Dialing DID DID is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant This service makes use of DID trunks which forward only the last three to
585. sertedID Web Tel to IP No Answer Timeout EMS IP Alert Timeout IPAlertTimeout Web Enable Remote Party ID EMS Enable RPI Header EnableRPlheader Version 6 0 6 Configuration Parameters Reference Description Determines whether the user phone string is added to the SIP URI and SIP To header 0 No user phone string is not added 1 Yes user phone string is part of the SIP URI and SIP To header default Determines whether the user phone string is added to the From and Contact SIP headers 0 No Doesn t add user phone string default 1 Yes user phone string is part of the From and Contact headers Determines the format of the URI in the P Asserted Identity and P Preferred Identity headers 0 Disable sip default 1 Enable tel Defines the time in seconds that the device waits for a 200 OK response from the called party IP side after sending an INVITE message If the timer expires the call is released The valid range is 0 to 3600 The default value is 180 Enables Remote Party Identity headers for calling and called numbers for Tel to IP calls 0 Disable default 1 Enable Remote Party Identity headers are generated in SIP INVITE messages for both called and calling numbers 249 March 2010 ca AudioCodes Parameter Web Enable History Info Header EMS Enable History Info EnableHistoryInfo Web Use Tgrp Information EMS U
586. set in the dotted decimal format in other words 192 168 0 0 16 is synonymous with 192 168 0 0 and a subnet 255 255 0 0 Refer to http en wikipedia org wiki Classless Inter Domain Routing for more information This CIDR notation lists the number of 1 bits in the subnet mask So a subnet mask of 255 0 0 0 when broken down to its binary format is represented by a prefix length of 8 11111111 00000000 00000000 00000000 and a subnet mask of 255 255 255 252 is represented by a prefix length of 30 11111111 11111111 11111111 11111100 Each interface must have its own address space Two interfaces may not share the same address space or even part of it The IP address should be configured as a dotted decimal notation For IPv4 interfaces the prefix length values range from 0 to 31 OAMP Interface Address when Booting using BootP DHCP When booting using BootP DHCP protocols an IP address is obtained from the server This address is used as the OAMP address for this session overriding the address configured using the Multiple Interface table The address specified for OAMP applications in the table becomes available when booting from flash again This allows the device to operate with a temporary address for initial management and configuration while retaining the address to be used for deployment 10 8 1 2 5Gateway Column This column defines a default gateway for the device For this reason only one default gateway may be configured
587. ss level of the secondary account a b From the Access Level drop down list select the new access level Click Change Access Level the new access level is applied immediately The access level of the primary Web user account is Security Administrator which cannot be modified The access level of the secondary account can only be modified by the primary account user or a secondary account user with Security Administrator access level 3 To change the user name of an account perform the following a b In the field User Name enter the new user name maximum of 19 case sensitive characters Click Change User Name if you are currently logged into the Web interface with this account the Enter Network Password dialog box appears requesting you to enter the new user name 4 To change the password of an account perform the following a b SIP User s Manual In the field Current Password enter the current password In the fields New Password and Confirm New Password enter the new password maximum of 19 case sensitive characters Click Change Password if you are currently logged into the Web interface with this account the Enter Network Password dialog box appears requesting you to enter the new password For security it s recommended that you change the default user name and password A Web user with access level Security Administrator can change all attributes
588. ss must be available on one of the local subnets The Interface column must be set to the Interface that the Gateway is configured on The Metric column must be set to 1 m The Routing Table configuration unlike the Multiple Interface table configuration is online Therefore the changes made to the routing rules are applied immediately Troubleshooting the Routing Table When adding or modifying any of the routing rules the added or modified rule passes a validation test If errors are found the route is rejected and is not added to the Routing table Failed routing validations may result in limited connectivity or no connectivity to the destinations specified in the incorrect routing rule For any error found in the Routing table or failure to configure a routing rule the device sends a notification message to the Syslog server reporting the problem Common routing rule configuration errors may include the following m The IP address specified in the Gateway column is unreachable from the interface specified in the Interface column m The same destination is defined in two different routing rules Subnet Mask and Prefix Length columns are both entered with inconsistent values and the Prefix Length overrides the Subnet Mask column m More than 25 routing rules were specified If a routing rule is required to access OAMP applications for remote management for instance and this route is not configured correc
589. ssword from its default value An example of a User Information file is shown in the figure below Figure 8 1 Example of a User Information File k UserInformationFile1000 txt Notepad i 5 xj 401 6380001 DN401 UN401 401 405 6380005 DN405 UN405 401 6 DN406 UN406 401 407 6380007 DN407 UN407 401 408 6380008 DN408 UN408 401 Note The last line in the User Information file must end with a carriage return i e by pressing the lt Enter gt key The User Information file can be loaded to the device by using one of the following methods mini file using the parameter UserlnfoFileName described in Auxiliary and Configuration Files Parameters on page 361 m Web interface refer to Loading Auxiliary Files on page 163 Automatic update mechanism using the parameter UserlInfoFileURL refer to the Product Reference Manual Each PBX extension registers separately a REGISTER message is sent for each entry only if AuthenticationMode is set to Per Endpoint using the IP number in the From To headers The REGISTER messages are sent gradually Initially the device sends requests according to the maximum number of allowed SIP dialogs configured by the parameter NumberOfActiveDialogs After each received response the subsequent request is sent Therefore no more than NumberOfActiveDialogs dialogs are active simultaneously The user name and password are used for SIP Authentication when required Version 6
590. st Hunt Groups e g an IP PBX to a Serving IP Group e g an Internet Telephony Service Provider ITSP The format of this parameter is as follows Account FORMAT Account_Index Account_ServedTrunkGroup Account ServedlPGroup Account ServinglPGroup Account Username Account Password Account HostName Account Register Account ContactUser Account ApplicationType Account For example Account 1 1 1 1 user 1234 acl 1 ITSP1 Notes This table can include up to 10 indices where 1 is the first index The parameter Account_ApplicationType is not applicable The parameter Account ServedlPGroup is not applicable You can define multiple table indices with the same ServedTrunkGroup but different ServinglPGroups username password HostName and ContactUser This provides the capability for registering the same Hunt Group to several ITSP s i e Serving IP Groups Fora detailed description of this table s parameters and for configuring this table using the Web interface refer to Configuring the Account Table on page 93 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Enables the use of a SIP Proxy server 0 No Proxy isn t used and instead the internal routing table is used default 1 Yes Proxy is used If you are using a Proxy server enter the IP address of the Proxy server in the Proxy Sets table
591. st Name Account_HostName Register Account_Register Contact User Account_ContactUser Application Type Account_ApplicationType Version 6 0 3 Web Based Management Description Digest MD5 Authentication password up to 50 characters Note After you click the Apply button this password is displayed as an asterisk Defines the Address of Record AOR host name It appears in REGISTER From To headers as ContactUser HostName For successful registrations this HostName is also included in the INVITE request s From header URI If not configured or if registration fails the SIP Group Name parameter from the IP Group table is used instead This parameter can be up to 49 characters Enables registration 0 No Don t register 1 Yes Enables registration When enabled the device sends REGISTER requests to the Serving IP Group In addition to activate registration you also need to set the parameter Registration Mode to Per Account in the Hunt Group Settings table for the specific Hunt Group The Host Name i e host name in SIP From To headers and Contact User user in From To and Contact headers are taken from this table upon a successful registration See the example below REGISTER sip xyz SIP 2 0 Via SIP 2 0 UDP 10 33 37 78 branch z9hG4bKac1397582418 From lt sip ContactUsereHostName gt tag 1c1397576231 To lt sip ContactUsereHostName gt Call ID 1397568957261200022256
592. st digit 0 Disable default 1 Enable Interworking of CPC is performed When this parameter is enabled the device sends the Caller ID number calling number with the cpc code received in the SIP INVITE message to the device s FXS port The cpc code is added as a prefix to the caller ID after IP to Tel calling number manipulation For example assuming that the incoming INVITE contains the following From or P Asserted Id header From lt sip 551 137077801 cpc payphone 10 20 7 35 gt tag 53700 The calling number manipulation removes 55 leaving 10 digits and then adds the prefix 7 the cpc code for payphone user Therefore the Caller ID number that is sent to the FXS port in this example is 71137077801 If the incoming INVITE message doesn t contain the cpc parameter nothing is added to the Caller ID number CPC Value in CPC Code Description Received INVITE Prefixed to Caller ID Sent to FXS Endpoint cpc unknown 1 Unknown user cpc subscribe 1 cpc ordinary 1 Ordinary user cpc priority 2 Pre paid user cpc test 3 Test user cpc operator 5 Operator cpc data 6 Data call 7 cpc payphone Payphone user Notes This parameter is applicable only to FXS interfaces For this parameter to be enabled you must also set the parameter EnableCallingPartyCategory to 1 Disables the generation of Caller ID type 2 when the phone is off hooked Caller ID type 2 also known as off hook Caller ID is sent to
593. status and the network interface m Reset button on the front panel for restarting the MP 124 and for restoring the MP 124 parameters to their factory default settings 1 3 SIP Overview Session Initiation Protocol SIP is an application layer control signaling protocol used on the gateway for creating modifying and terminating sessions with one or more participants These sessions can include Internet telephone calls media announcements and conferences SIP invitations are used to create sessions and carry session descriptions that enable participants to agree on a set of compatible media types SIP uses elements called Proxy servers to help route requests to the user s current location authenticate and authorize users for services implement provider call routing policies and provide features to users SIP also provides a registration function that enables users to upload their current locations for use by Proxy servers SIP implemented in the gateway complies with the Internet Engineering Task Force IETF RFC 3261 refer to http www ietf org Version 6 0 19 March 2010 A ll AudioCodes MediaPack Series Reader s Notes SIP User s Manual 20 Document LTRT 65413 SIP User s Manual 2 Configuration Concepts 2 Configuration Concepts You can configure the device using the following management tools m The device s HTTP based Embedded Web Server Web interface using any standard Web browser described in Web b
594. t disconnect detection is considered The valid range is 0 to 20 Volts The default value is 4 Volts Notes This parameter is applicable only to FXO interfaces For this parameter to take effect a device reset is required Determines the frequency at which the analog line voltage is sampled after offhook for detection of the current disconnect threshold The valid range is 100 to 2500 msec The default value is 1000 msec Notes This parameter is applicable only to FXO interfaces For this parameter to take effect a device reset is required 317 March 2010 7a tal AudioCodes MediaPack Series 6 11 Tone Parameters This subsection describes the device s tone parameters 6 11 1 Telephony Tone Parameters The telephony tone parameters are described in the table below Table 6 48 Tone Parameters Parameter Description Tone Index Table Tonelndex This ini file table parameter configures the Tone Index table which allows you to define Distinctive Ringing and Call Waiting tones per FXS endpoint or for a range of FXS endpoints and is based on calling number source number prefix for IP to Tel calls This allows different tones to be played for an FXS endpoint depending on the source number of the received call The format of this parameter is as follows Tonelndex FORMAT Tonelndex_Index Tonelndex_FXSPort_First Tonelndex_FXSPort_Last Tonelndex_SourcePrefix Tonelndex Prioritylndex
595. t enable or disable the silence suppression option for the selected coder 7 Repeat steps 2 through 6 for the next optional coders 8 Click the Submit button to save your changes 9 To save the changes to flash memory refer to Saving Configuration on page 161 A coder i e Coder Name can appear only once in the table If packetization time and or rate are not specified the default value is applied Only the packetization time of the first coder in the coder list is declared in INVITE 200 OK SDP even if multiple coders are defined For G 729 it s also possible to select silence suppression without adaptations If the coder G 729 is selected and silence suppression is disabled for this coder the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is enabled or set to Enable w o Adaptations annexb yes is included An exception to this logic is when the remote gateway is a Cisco device IsCiscoSCEMode Version 6 0 103 March 2010 Ao L tal AudioCodes MediaPack Series 3 3 4 5 2 Configuring Coder Groups The Coder Group Settings page provides a table for defining up to four different coder groups These coder groups are used in the Tel Profile Settings and IP Profile Settings pages to assign different coders to Profiles For each coder group you can define up to ten coders where the first coder and its attributes in the table takes precede
596. t be defined by User Cloning The SNMP Manager creates a new user according to the original user permission levels gt 1 To clone SNMPv3 Users Open the SNMPv3 Users screen Configuration icon gt Network Frame menu gt SNMPv3 Users tab Select the user with which you wish to clone permission levels Click the lal button the New SNMPv3 User window appears Provide a new user name old passwords of the user you clone permissions from and new user passwords Select a User permission group If the new user wishes to receive traps to the user defined destination select the Use SNMPv3 User Security Profile for Trap Forwarding option to provision Trap destination IP and Port EMS adds this new user to the SNMP Trap Managers Table It is also possible to define an additional trap destination after a new user is defined Resetting the Device When you have completed configuring the device you need to save your settings to the device s flash memory and reset the device gt 1 2 To save configuration and reset the device In the MG Tree select the device that you want to reset On the Actions bar click the Reset H button Figure 5 10 Confirmation for Saving Configuration and Resetting Device 3 4 5 Question Ensure that the option Burn Configuration into flash memory is selected Click Yes the progress of the reset process is displayed Click Done when complete SIP User s Manual 204 Document
597. t from the value of the parameter TCPLocalSIPPort 247 March 2010 ca AudioCodes Parameter Web EMS Enable SIPS EnableSIPS Web EMS Enable TCP Connection Reuse EnableTCPConnectionReuse Web EMS Reliable Connection Persistent Mode ReliableConnectionPersistent Mode Web EMS TCP Timeout SIPTCPTimeout Web SIP Destination Port EMS Destination Port SIPDestinationPort SIP User s Manual MediaPack Series Description Enables secured SIP SIPS URI connections over multiple hops 0 Disable default 1 Enable When the parameter SIPTransportType is set to 2 i e TLS and the parameter EnableSIPS is disabled TLS is used for the next network hop only When the parameter SIPTransportType is set to 2 or 1 i e TCP or TLS and EnableSIPS is enabled TLS is used through the entire connection over multiple hops Note If this parameter is enabled and the parameter SIPTransportType is set to 0 i e UDP the connection fails Enables the reuse of the same TCP connection for all calls to the same destination 0 Disable Use a separate TCP connection for each call 1 Enable Use the same TCP connection for all calls default Determines whether all TCP TLS connections are set as persistent and therefore not released 0 Disable default all TCP connections except those that are set to a proxy IP are released if not used by any SIP dialog transaction 1 Enable
598. tType you can configure whether to pass V 34 over T38 fax relay or use Bypass over the High Bit Rate coder e g PCM A Law Note The CNG detector is disabled CNGDetectorMode 0 in all the subsequent examples 9 6 3 1 Using Bypass Mechanism for V 34 Fax Transmission In this proprietary scenario the device uses bypass or NSE mode to transmit V 34 faxes enabling the full utilization of its speed Configure the following parameters to use bypass mode for both T 30 and V 34 faxes m FaxTransportMode 2 Bypass m V34ModemTransportType 2 Modem bypass SIP User s Manual 406 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities m V32ModemTransportType 2 m V23ModemTransportType 2 m V22ModemTransportType 2 Configure the following parameters to use bypass mode for V 34 faxes and T 38 for T 30 faxes m FaxTransportMode 1 Relay V34ModemTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemTransportType 2 V22ModemTransportType 2 9 6 3 2 Using Relay mode for both T 30 and V 34 faxes In this scenario V 34 fax machines are forced to use their backward compatibility with T 30 faxes and operate in the slower T 30 mode Use the following parameters to use T 38 mode for both V 34 and T 30 faxes Version 6 0 FaxTransportMode 1 Relay V34ModemTransportType 0 Transparent V32ModemTransportType 0 V23ModemTransportType 0 V22ModemTransportType 0 407 March 2010 A
599. table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 23 Call Forward Table Parameter Description Forward Type Determines the scenario for forwarding a call 0 Deactivate Don t forward incoming calls default 1 On Busy Forward incoming calls when the port is busy 2 Unconditional Always forward incoming calls 3 No Answer Forward incoming calls that are not answered within the time specified in the Time for No Reply Forward field 4 On Busy or No Answer Forward incoming calls when the port is busy or when calls are not answered within the time specified in the Time for No Reply Forward field 5 Do Not Disturb Immediately reject incoming calls Forward to Phone The telephone number or URI lt number gt A lt IP address gt to where the Number call is forwarded SIP User s Manual 140 Document LTRT 65413 SIP User s Manual 3 Web Based Management Parameter Description Note If this field only contains a telephone number and a Proxy isn t used the forward to phone number must be specified in the Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 Time for No Reply If you have set the Forward Type for this port to No Answer enter the Forward number of seconds the device waits before forwarding the call to the phone number specified 3 3 4
600. tact sip 122 10 1 1 200 Expires 3600 Authorization Digest username 122 realm audiocodes com nonce 11432d6bce58ddf02e3b5e1c77c010d2 WiPte7 10 2 2 222 response b9c45d0234ababf5ddf5c704029b38cf 7 Upon receiving this request and if accepted by the Proxy the proxy returns a 200 OK response closing the REGISTER transaction SIP 2 0 200 OK Via SIP 2 0 UDP 10 1 1 200 From lt sip 122 10 1 1 200 gt tag 1c23940 TORS Sip 12200 i i1 200 Call ID 654982194 10 1 1 200 Cseg 1 REGISTER Date Thu 26 Jul 2001 09 34 42 GMT Server Columbia SIP Server 1 17 Content Length 0 Contact lt sip 122 10 1 1 200 gt expires Thu 26 Jul 2001 10 34 42 GMT action proxy q 1 00 Contact lt 122 10 1 1 200 gt expires Tue 19 Jan 2038 03 14 07 GMT action proxy q 0 00 Expires Thu 26 Jul 2001 10 34 42 GMT Version 6 0 425 March 2010 A tal AudioCodes MediaPack Series 9 8 3 Proxy or Registrar Registration Example Below is an example of Proxy and Registrar registration REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z9hG4bRaC7AU234 From lt sip GWRegistrationNameGsipgatewayname gt tag 1c29347 To lt sip GWRegistrationName sipgatewayname gt Cali TDg 1045530212 17922 229 Seq 1 REGISTER Expires 3600 Contact sip GWRegistrationName 212 179 22 229 Content Length 0 The servername string is defined acc
601. ted is overridden by the parameter SourceNumberMaplp2Tel IsPresentationRestricted in the Source Number Manipulation table table parameter SourceNumberMapIP2Tel For configuring this table using the Web interface refer to Configuring Caller Display Information on page 138 Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 290 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Web EMS Enable Caller ID EnableCallerlD Web Caller ID Type EMS Caller id Types CallerIDType Version 6 0 Description Determines whether Caller ID is enabled 0 Disable Disable the Caller ID service default 1 Enable Enable the Caller ID service If the Caller ID service is enabled then for FXS interfaces calling number and Display text from IP are sent to the device s port For FXO interfaces the Caller ID signal is detected and sent to IP in the SIP INVITE message as Display element For information on the Caller ID table refer to Configuring Caller Display Information on page 138 To disable enable caller ID generation per port refer to Configuring Call Forward on page 140 Defines one of the following standards for detection FXO and generation FXS of Caller ID and detection FXO generation FXS of MWI when specified signals 0 Standard Bellcore Caller ID a
602. ters The Ethernet parameters are described in the table below Table 6 1 Ethernet Parameters Parameter Description EMS Physical Configuration Defines the Ethernet connection mode type EthernetPhyConfiguration 9 10Base T half duplex Not applicable 1 10Base T full duplex 2 100Base TX half duplex 3 100Base TX full duplex 4 Auto negotiate default For detailed information on Ethernet interface configuration refer to Ethernet Interface Configuration on page 443 Note For this parameter to take effect a device reset is required Web 802 1x Mode Enables support for IEEE 802 1x physical port security The device EMS Mode can function as an IEEE 802 1X supplicant IEEE 802 1X is a 802 1xMode standard for port level security on secure Ethernet switches when a unit is connected to a secure port no traffic is allowed until the identity of the unit is authenticated 0 Disabled default 1 EAP MD5 Authentication is performed using a user name and password configured by the parameters 802 1xUsername and 802 1xPassword 2 Protected EAP Authentication is performed using a user name and password configured by the parameters 802 1xUsername and 802 1xPassword In addition the protocol used is MSCHAPv2 over an encrypted TLS tunnel 3 EAP TLS The device s certificate is used to establish a mutually authenticated TLS session with the Access Server This Version 6 0 207 March
603. th a 128 bit key and HMAC SHA1 message authentication with a 80 bit tag 21 AES CM 128 HMAC SHAT1 32 device uses AES CM encryption with a 128 bit key and HMAC SHA1 message authentication with a 32 bit tag On a secured RTP session this parameter determines whether to enable authentication on transmitted RTP packets 0 Enable default 1 Disable On a secured RTP session this parameter determines whether to enable encryption on transmitted RTP packets 0 Enable default 1 Disable On a secured RTP session this parameter determines whether to enable encryption on transmitted RTCP packets 0 Enable default 1 Disable 235 March 2010 ca AudioCodes 6 4 4 TLS Parameters MediaPack Series The Transport Layer Security TLS parameters are described in the table below Parameter Web EMS TLS Version TLSVersion Web TLS Client Re Handshake Interval EMS TLS Re Handshake Interval TLSReHandshakelnterval Web TLS Mutual Authentication EMS SIPS Require Client Certificate SIPSRequireClientCertificate Web EMS Peer Host Name Verification Mode PeerHostNameVerificationMode SIP User s Manual Table 6 21 TLS Parameters Description Defines the supported versions of SSL TLS Secure Socket Layer Transport Layer Security 0 SSL 2 0 3 0 and TLS 1 0 SSL 2 0 SSL 3 0 and TLS 1 0 are supported default 4 TLS 1 0 Only only TLS 1 0 is used When set to 0
604. th the same NPI TON or Phone Context are allowed In this scenario a Tel to IP call uses the first match Phone Context is unique in that it doesn t appear in the Request URI as a Phone Context parameter Instead it s added as a prefix to the phone number The isn t removed from the phone number in the IP to Tel direction To configure the Phone Context table using the Web interface refer to Mapping NPI TON to SIP Phone Context on page 122 Fora description on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Determines whether the Hunt Group ID is added as a prefix to the destination phone number i e called number for Tel to IP calls 0 No Don t add Hunt Group ID as prefix default 1 Yes Add Hunt Group ID as prefix to called number Notes This option can be used to define various routing rules To use this feature you must configure the Hunt Group IDs refer to Configuring the Endpoint Phone Numbers on page 143 343 March 2010 ca AudioCodes Parameter Web Add Trunk ID as Prefix EMS Add Port ID As Prefix AddPortAsPrefix Web EMS Add Trunk Group ID as Prefix to Source AddTrunkGroupAsPrefixToSource Web IP to Tel Remove Routing Table Prefix EMS Remove Prefix RemovePrefix SwapTel2IPCalled amp CallingNumbers Web EMS Source Manipulation Mode SourceManipulationMode SIP User s Manual MediaPack Series
605. the Subject Name field enter the DNS name and then click Generate CSR A textual certificate signing request that contains the SSL device identifier is displayed Copy this text and send it to your security provider The security provider also known as Certification Authority or CA signs this request and then sends you a server certificate for the device Save the certificate to a file e g cert txt Ensure that the file is a plain text file containing the BEGIN CERTIFICATE header as shown in the example of a Base64 Encoded X 509 Certificate below MIIDkzCCAnugAwIBAgIEAgAAADANBgkghkiG9w0BAOOFADA MOswCOYDVOOGEWJGUj ETMBEGA1UEChMKO2VydGlwb3N0ZTEDMBkGA1UBAxMSO2VydGlwb3N0ZSBTZXJ2ZXVy MB4XDTk4MDYyNDA4MDAwMFoXDTE4MDYyNDA4MDAwMFowPzELMAkGA1UEBhMCR1IxEz ARBGNVBAOTCKN1 cnRpcG9 zdGUxGzZAZBgNVBAMTEKN1 cnRpcG9 zdGUgU2VydmV1cjCC ASEwDOYJKoZIhvcNAOEBBOADggEOADCCAOkCggEAPgd4MziR4spWldGRx8borhZkon WnNm Yhb7 4067ecf1janH7GcN SXsfx7jJpreWULE7v7Cvpr4R7gIJcmdHIntmf7 JPM5n6cDBv17uSW63er7NkVnMFHwK10aGFLMybFkzaeGrvFm4k31RefiXDmuOe FhJ gHYezYHf44LvPRPwhSrzi9 Aq308pWDguJuZDIUP1F1jMa LPwvREXf FcUW w In the Certificates Files group click the Browse button corresponding to Send Server Certificate navigate to the cert txt file and then click Send File When the loading of the certificate is complete save the configuration refer to Saving Configuration on page 161 and restart the device the Web interface uses the provided certificate
606. the IP connection is disallowed m DNS resolution When host name is used instead of IP address for the destination route it is resolved to an IP address by a DNS server Connectivity and QoS are then applied to the resolved IP address Version 6 0 399 March 2010 7a e AudioCodes MediaPack Series 9 6 9 6 1 9 6 2 Fax and Modem Capabilities This section describes the device s fax and modem capabilities and includes the following main subsections m Fax and modem operating modes refer to Fax Modem Operating Modes on page 400 m Fax and modem transport modes refer to Fax Modem Transport Modes on page 400 V 34 fax support refer to V 34 Fax Support on page 406 m V 152 support refer to V 152 Support on page 408 Fax Modem Operating Modes The device supports two modes of operation m Fax modem negotiation that is not performed during the establishment of the call m Voice band data VBD mode for V 152 implementation refer to V 152 Support on page 408 fax modem capabilities are negotiated between the device and the remote endpoint at the establishment of the call During a call when a fax modem signal is detected transition from voice to VBD or T 38 is automatically performed and no additional SIP signaling is required If negotiation fails i e no match is achieved for any of the transport capabilities fallback to existing logic occurs according to the parameter IsFaxUsed Fax Modem Tr
607. the cmp and Software Upgrade Key files are loaded to the device Note To load the Software Upgrade Key using BootP TFTP the extension name of the key file must be ini Software Upgrade Wizard The Software Upgrade Wizard allows you to upgrade the device s firmware cmp file as well as load an ini file and or auxiliary files e g Call Progress Tones However it is mandatory when using the wizard to first load a cmp file to the device You can then choose to also load an ini file and or auxiliary files but this cannot be pursued without first loading a cmp file For the ini and each auxiliary file type you can choose to load a new file or not load a file but use the existing file i e maintain existing configuration running on the device The Software Upgrade Wizard allows you to load the following files cmp Mandatory compressed firmware file Optional files e ini configuration file e Auxiliary files CPT Call Progress Tone PRT Prerecorded Tones and USERINF User Information Warnings e To preserve all configuration settings before upgrading the device to a new major software version e g from version 5 8 to 6 0 save a copy of the device s configuration settings i e ini file to your PC and ensure that you have all the original auxiliary files currently used by the device After you have upgraded the device restore your configuration settings by uploading these files to the device For savi
608. the device s CLI refer to Configuring the Web User Accounts on page 66 222 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 2 2 Web Parameters The Web parameters are described in the table below Parameter DisableWebTask HTTPport EMS Disable WEB Config DisableWebConfig ResetWebPassword ScenarioFileName Version 6 0 Table 6 11 Web Parameters Description Disables or enables device management through the Web interface 0 Enable Web management default 1 Disable Web management Note For this parameter to take effect a device reset is required HTTP port used for Web management default is 80 Note For this parameter to take effect a device reset is required Determines whether the entire Web interface is in read only mode 0 Enables modifications of parameters default 1 Web interface in read only mode When in read only mode parameters can t be modified In addition the following pages can t be accessed Web User Accounts Certificates Regional Settings Maintenance Actions and all file loading pages Load Auxiliary Files Software Upgrade Wizard and Configuration File Notes For this parameter to take effect a device reset is required To return to read write after you have applied read only using this parameter set to 1 you need to reboot your device with an ini file that doesn t in
609. the gateway parameter NSEPayloadType 100 In NSE bypass mode the device starts using G 711 A Law default or G 711u Law according to the parameter FaxModemBypassCoderType The payload type used with these G 711 coders is a standard one 8 for G 711 A Law and 0 for G 711 p Law The parameters defining payload type for the old Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass The bypass packet interval is selected according to the parameter FaxModemBypassBasicRtpPacketInterval NSE payload type for Cisco Bypass compatible mode The valid range is 96 127 The default value is 105 Note Cisco gateways usually use NSE payload type of 100 V 21 Modem Transport Type used by the device 0 Disable Disable Transparent default 1 Enable Relay N A 2 Enable Bypass 3 Events Only Transparent with Events 354 Document LTRT 65413 SIP User s Manual Parameter Web V 22 Modem Transport Type EMS V22 Transport V22ModemTransportType Web V 23 Modem Transport Type EMS V23 Transport V23ModemTransportType Web V 32 Modem Transport Type EMS V32 Transport V32ModemTransportType Web V 34 Modem Transport Type EMS V34 Transport V34ModemTransportType EMS Bell Transport Type BellModemTransportType Version 6 0 6 Configuration Parameters Reference Description V 22 Modem Transport Type used by the device 0 Disable Disable Transparent 1 E
610. the outgoing SDP s c field to 0 0 0 0 default 1 Sets the IP address of the outgoing SDP c field to the IP address of the device If the incoming SDP doesn t contain the a inactive line the returned SDP contains the a recvonly line 255 March 2010 ca AudioCodes Parameter Web EMS Enable Delayed Offer EnableDelayedOffer Web EMS Enable Contact Restriction EnableContactRestriction AnonymousMode EMS P Asserted User Name PAssertedUserName EMS Use URL In Refer To Header UseAORInReferToHeader Web Enable User Information Usage EnableUserlnfoUsage SIP User s Manual MediaPack Series Description Determines whether the device sends the initial INVITE message with or without an SDP Sending the first INVITE without SDP is typically done by clients for obtaining the far end s full list of capabilities before sending their own offer An alternative method for obtaining the list of supported capabilities is by using SIP OPTIONS which is not supported by every SIP agent 0 Disable The device sends the initial INVITE message with an SDP default 1 Enable The device sends the initial INVITE message without an SDP Determines whether the device sets the Contact header of outgoing INVITE requests to anonymous for restricted calls 0 Disable default 1 Enable Determines whether the device s IP address is used as the URI host part instead of anonymous i
611. the parameter DNSQueryType is disabled Determines whether the device uses its IP address or gateway name in keep alive SIP OPTIONS messages 0 No Use the device s IP address in keep alive OPTIONS messages default 1 Yes Use Gateway Name SIPGatewayName in keep alive OPTIONS messages The OPTIONS Reguest URI host part contains either the device s IP address or a string defined by the parameter SIPGatewayName The device uses the OPTIONS request as a keep alive message to its primary and redundant Proxies i e the parameter EnableProxyKeepAlive is set to 1 User name used for Registration and Basic Digest authentication with a Proxy Registrar server The default value is an empty string Notes This parameter is applicable only if single device registration is used i e the parameter AuthenticationMode is set to authentication per gateway Instead of configuring this parameter the Authentication table can be used refer to Authentication on page 136 268 Document LTRT 65413 SIP User s Manual Parameter Web EMS Password Password Web EMS Cnonce Cnonce Web EMS Mutual Authentication Mode MutualAuthenticationMode Web EMS Challenge Caching Mode SIPChallengeCachingMode Version 6 0 6 Configuration Parameters Reference Description The password used for Basic Digest authentication with a Proxy Registrar server A single password is used for all device ports
612. the possible values of this column and their descriptions Table 10 2 Application Types Value Description 0 OAMP only OAMP applications are allowed on this interface 1 MEDIA only Media RTP are allowed on this interface 2 CONTROL only Call Control applications are allowed on this interface 3 OAMP amp MEDIA only OAMP and Media RTP applications are allowed on this interface 4 OAMP amp CONTROL only OAMP and Call Control applications are allowed on this interface 5 MEDIA 8 CONTROL only Media RTP and Call Control applications are allowed on this interface OAMP MEDIA 8 CONTROL all of the application types are allowed on this interface For valid configuration guidelines refer to Multiple Interface Table Configuration Summary and Guidelines on page 455 for more information 10 8 1 2 3Interface Mode Column The Interface Mode column determines the method that this interface uses to acquire its IP address For IPv4 Manual IP Address assignment use IPv4 Manual 10 SIP User s Manual 450 Document LTRT 65413 SIP User s Manual 10 Networking Capabilities 10 8 1 2 4IP Address and Prefix Length Columns These columns allow the user to configure an IPv4 IP address and its related subnet mask The Prefix Length column holds the Classless Inter Domain Routing CIDR style representation of a dotted decimal subnet notation The CIDR style representation uses a suffix indicating the number of bits which are
613. tification message is displayed SIP User s Manual 36 Document LTRT 65413 SIP User s Manual 3 Web Based Management 3 1 8 Working with Scenarios The Web interface allows you to create your own menu with up to 20 pages selected from the menus in the Navigation tree i e pertaining to the Configuration Management and Status amp Diagnostics tabs The menu is a set of configuration pages grouped into a logical entity referred to as a Scenario Each page in the Scenario is referred to as a Step For each Step you can select up to 25 parameters in the page that you want available in the Scenario Therefore the Scenario feature is useful in that it allows you quick and easy access to commonly used configuration parameters specific to your network environment When you login to the Web interface your Scenario is displayed in the Navigation tree thereby facilitating your configuration Instead of creating a Scenario you can also load an existing Scenario from a PC to the device refer to Loading a Scenario to the Device on page 42 3 1 8 1 Creating a Scenario The Web interface allows you to create one Scenario with up to 20 configuration pages as described in the procedure below gt To create a Scenario 1 On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm creation of a Scenario Figure 3 14 Scenario Creation Confirm Message Box Microsoft Internet Explorer A Creat
614. tion fields refer to the Product Reference Manual The BootP TFTP configuration utility displays this information in the Client Info column Notes For this parameter to take effect a device reset is required This option is not available on DHCP servers Version 6 0 231 March 2010 7a c tal AudioCodes MediaPack Series 6 4 Security Parameters This subsection describes the device s security parameters 6 4 1 General Parameters The general security parameters are described in the table below Parameter Web Voice Menu Password VoiceMenuPassword EnableSecureStartup Table 6 18 General Security Parameters Description The password for accessing the device s voice menu for configuration and status To activate the menu connect a POTS telephone and dial three stars followed by the password The default value is 12345 For detailed information on the voice menu refer to the device s Installation Manual Enables the Secure Startup mode In this mode downloading the ini file to the device is restricted to a URL provided in initial configuration see the parameter IniFileURL or using DHCP 0 Disable default 1 Enable disables TFTP and allows secure protocols such as HTTPS to fetch the device configuration For a detailed explanation on Secure Startup refer to the Product Reference Manual Note For this parameter to take effect a device reset is required Web Internal F
615. tion menu gt Endpoint Settings submenu gt Authentication page item Figure 3 84 Authentication Page Gateway Port User Name Password Port 1 FXS Port 2 FXS Port 3 FXS Port 4 FXS Pot 5 FRO Port6 FXO Port 7 FRO Port 8 FXO 3 In the User Name and Password fields corresponding to a port enter the user name and password respectively 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 161 3 3 4 9 2 Configuring Automatic Dialing The Automatic Dialing page allows you to define a telephone number that is automatically dialed when an FXS or FXO port is used e g off hooked After a ring signal is detected on an Enabled FXO port the device initiates a call to the destination number without seizing the line The line is seized only after the call is answered After a ring signal is detected on a Disabled or Hotline FXO port the device seizes the line You can also configure automatic dialing using the ini file table parameter TargetOfChannel You can configure the device to play a Busy Reorder tone to the Tel side upon receiving a SIP 4xx 5xx or 6xx response from the IP side i e Tel to IP call failure using the ini file parameter FXOAutoDialPlayBusyTone refer to SIP Configuration Parameters on page 245 Versi
616. tion time of the first coder in the defined coder list is declared in INVITE 200 OK SDP even if multiple coders are defined The device always uses the packetization time requested by the remote side for sending RTP packets If not specified the packetization time is assigned the default value The value of several fields is hard coded according to common standards e g payload type of G 711 U law is always 0 Other values can be set dynamically If no value is specified for a dynamic field a default value is assigned If a value is specified for a hard coded field the value is ignored f silence suppression is not defined for a specific coder the value defined by the parameter EnableSilenceCompression is used If G 729 is selected and silence suppression is enabled for this coder the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is set to Enable w o Adaptations annexb yes is included An exception is when the remote device is a Cisco gateway IsCiscoSCEMode The coder G 722 provides Packet Loss Concealment PLC capabilities ensuring higher voice quality Foran explanation on V 152 support and implementation of T 38 and VBD coders refer to V 152 Support on page 408 For a description of using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web IP Profile Settings Table EMS Protocol Definition gt
617. tionDisableTx Web Disable Encryption On Transmitted RTP Packets EMS RTP EncryptionDisable Tx RTPEncryptionDisableTx Web Disable Encryption On Transmitted RTCP Packets EMS RTCP EncryptionDisable Tx RTCPEncryptionDisableTx Version 6 0 6 Configuration Parameters Reference Description Determines the device s mode of operation when SRTP is used i e when the parameter EnableMediaSecurity is set to 1 0 Preferable The device initiates encrypted calls If negotiation of the cipher suite fails an unencrypted call is established Incoming calls that don t include encryption information are accepted default 1 Mandatory The device initiates encrypted calls but if negotiation of the cipher suite fails the call is terminated Incoming calls that don t include encryption information are rejected 2 Preferable Single Media The device sends SDP with only a single media m line e g m audio 6000 RTP AVP 4 0 70 96 with RTP AVP and crypto keys If the remote SIP UA does not support SRTP it ignores the crypto lines Note Before configuring this parameter set the parameter EnableMediaSecurity parameter to 1 Determines the size in bytes of the Master Key Identifier MKI in SRTP Tx packets The range is 0 to 4 The default value is 0 Defines the offered SRTP crypto suites 0 All All available crypto suites default 1TAES CM 128 HMAC SHAT1 80 device uses AES CM encryption wi
618. tistics menu allows you to monitor real time activity such as IP connectivity information call details and call statistics including the number of call attempts failed calls fax calls etc This menu includes the following page items IP to Tel Calls Count refer to Viewing Call Counters on page 178 Tel to IP Calls Count refer to Viewing Call Counters on page 178 SAS Registered Users refer to Viewing SAS Registered Users on page 180 Call Routing Status refer to Viewing Call Routing Status on page 181 Registration Status refer to Viewing Registration Status on page 181 IP Connectivity refer to Viewing IP Connectivity on page 183 Note The Web pages pertaining to the Gateway Statistics menu do not refresh automatically To view updated information close the relevant page and then re access it Viewing Call Counters The IP to Tel Calls Count and Tel to IP Calls Count pages provide you with statistical information on incoming IP to Tel and outgoing Tel to IP calls The statistical information is updated according to the release reason that is received after a call is terminated during the same time as the end of call Call Detail Record or CDR message is sent The release reason can be viewed in the Termination Reason field in the CDR message You can reset the statistical data displayed on the page i e refresh the display by clicking the Reset Counters button located on the page
619. tivity Log to a Syslog server for reporting certain types of Web operations Messages according to the below user defined filters ActivityListToLog PVC Parameters Value Change Changes made on the fly to parameters AFL Auxiliary Files Loading Loading of auxiliary files DR Device Reset Reset of device via the Maintenance Actions page FB Flash Memory Burning Burning of files or parameters to flash in Maintenance Actions page SWU Device Software Update cmp file loading via the Software Upgrade Wizard ARD Access to Restricted Domains Access to restricted domains which include the following Web pages 1 ini parameters AdminPage General Security Settings Configuration File IPSec IKE tables Software Upgrade Key Internal Firewall Web Access List 8 Web User Accounts NAA Non Authorized Access Attempt to access the Web interface with a false or empty user name or password 2 3 4 5 6 7 KARLARLA SPC Sensitive Parameters Value Change Changes made to sensitive parameters v 1 IP Address v 2 Subnet Mask v 3 Default Gateway IP Address v 4 ActivityListToLog For example ActivityListToLog pvc afl dr fb swu ard naa spc SIP User s Manual 228 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference 6 3 3 Remote Alarm Indication Parameters The Re
620. tly the route is not added and the device is not accessible remotely To restore connectivity the device must be accessed locally from the OAMP subnet and the required routes be configured Version 6 0 459 March 2010 c tal AudioCodes MediaPack Series 10 8 3 Setting up the Device 10 8 3 1 Using the Web Interface The Web interface is a convenient user interface for configuring the device s network configuration 10 8 3 2 Using the ini File When configuring the network configuration using the ini File use a textual presentation of the Interface and Routing Tables as well as some other parameters The following shows an example of a full network configuration consisting of all the parameters described in this section VLAN related parameters VlanMode 0 VlanNativeVlanId 1 Routing Table Configuration RoutingTableDestinationsColumn 201 201 0 0 202 202 0 0 RoutingTableDestinationPrefixLensColumn 16 16 RoutingTableGatewaysColumn 192 168 0 2 192 168 0 3 RoutingTableInterfacesColumn 0 0 RoutingTableHopsCountColumn 1 1 Class Of Service parameters VlanNetworkServiceClassPriority 7 VlanPremiumServiceClassMediaPriority 6 VlanPremiumServiceClassControlPriority 6 VlanGoldServiceClassPriority 4 VlanBronzeServiceClassPriority 2 NetworkServiceClassDiffServ 48 PremiumServiceClassMediaDiffServ 46 PremiumServiceClassControlDiffServ 40 GoldServiceClassDiffServ 26 BronzeServiceClas
621. tnPrefix 0 100 1 200 0 2 PstnPrefix 1 2 1 3 acl joe Notes This parameter can include up to 24 indices Fora description of the table s parameters refer to the corresponding Web parameters in Configuring the IP to Hunt Group Routing Table on page 131 To support the In Call Alternative Routing feature you can use two entries that support the same call but assigned with a different Hunt Group The second entry functions as an alternative route if the first rule fails as a result of one of the release reasons configured in the AltRouteCauselP2Tel table Selection of Hunt Groups for IP to Tel calls is according to destination number source number and source IP address The source IP address SourceAddress can include the x wildcard to represent single digits For example 10 8 8 xx represents IP addresses between 10 8 8 10 and 10 8 8 99 The source IP address SourceAddress can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 If the source IP address SourceAddress includes an FQDN DNS resolution is performed according to the parameter DNSQueryType For available notations for depicting a range of multiple numbers refer to Dialing Plan Notation for Routing and Manipulation on page 377 Fora description on using ini file table parameters refer to Configuring ini Fi
622. to challenge authentication containing a WWW Authenticate header and expect the re INVITE to contain an Authorization header Version 6 0 423 March 2010 A c tal AudioCodes MediaPack Series The following example describes the Digest Authentication procedure including computation of user agent credentials 1 The REGISTER request is sent to a Registrar Proxy server for registration REGISTER sip 10 2 2 222 SIP 2 0 Via SIP 2 0 UDP 10 1 1 200 From lt Sip 122 10 1 1 200 gt tag 1c17940 Wes zemos UAA 10 i 1 200s Call ID 634293194 10 1 1 200 User Agent Audiocodes Sip Gateway MediaPack v 6 00 010 006 CSeq 1 REGISTER Contact sip 122 10 1 1 200 Expires 3600 2 Upon receipt of this request the Registrar Proxy returns a 401 Unauthorized response SIP 2 0 401 Unauthorized Via SIP 2 0 UDP 10 2 1 200 priom lt sijoglAA LO 2 2 822 Ca le nA Mog SPO P220M TRPE Calil mD S342931 400 11200 Cseg 1 REGISTER Date Mon 30 Jul 2001 15 33 54 GMT Server Columbia SIP Server 1 17 Content Length 0 WWW Authenticate Digest realm audiocodes com nonce 11432d6bce58daf02e3b5e1c77c010d2 stale FALSE algorithm MD5 3 According to the sub header present in the WWW Authenticate header the correct REGISTER request is created 4 Since the algorithm is MD5 e The username is equal to the endpoint phone number 122 e The realm return by the proxy is audiocodes com e The password from th
623. to all numbers that start with 55 including numbers that start with 551 Number manipulation can occur before or after a routing decision is made For example you can route a call to a specific Hunt Group according to its original number and then you can remove or add a prefix to that number before it is routed To determine when number manipulation is performed configure the IP to Tel Routing Mode parameter RouteModelP2Tel described in Configuring the IP to Hunt Group Routing Table on page 131 and Tel to IP Routing Mode parameter RouteModeTel2IP described in Configuring the Tel to IP Routing on page 126 The manipulation rules are executed in the following order Stripped digits from left Stripped digits from right Number of digits to leave Prefix to add Suffix to add SIP User s Manual 116 Document LTRT 65413 SIP User s Manual 3 Web Based Management Number manipulation can occur before or after a routing decision is The manipulation rules can be applied to any incoming call whose source IP address if applicable source Hunt Group if applicable source IP Group if applicable destination number prefix and source number prefix matches the values defined in the Source IP Address Source Trunk Group Source IP Group Destination Prefix and Source Prefix fields respectively The number manipulation can be performed using a combination of each of the above criteria or using each
624. tring in their names are listed gt 1 To search for ini file parameters configurable in the Web interface On the Navigation bar click the Search tab the Search engine appears in the Navigation pane In the Search field enter the parameter name or sub string of the parameter name that you want to search If you have performed a previous search for such a parameter instead of entering the required string you can use the Search History drop down list to select the string saved from a previous search Click Search a list of located parameters based on your search appears in the Navigation pane Each searched result displays the following e __ ini file parameter name e Link in green to its location page in the Web interface e Brief description of the parameter In the searched list click the required parameter link in green to open the page in which the parameter appears the relevant page opens in the Work pane and the searched parameter is highlighted for easy identification as shown in the figure below Figure 3 13 Searched Result Screen Contiguration Management Pisgnostice sommano Search Parameter Highlighted in Page Basic Full Search field 5 Search History VLANDRONZESERVICECLASSPRIOR A Links VL a Searched Results identifier VLANNE ORK SERVICECLASSPRI Link VLANS etting Sets the priority for Network service Note Ifthe searched parameter is not located a no
625. ts for the second cadence on off cycle e Second Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the second cadence on off cycle e Third Burst Ring On Time 10 msec Ring On period in 10 msec units for the third cadence on off cycle e Third Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the third cadence on off cycle e Fourth Burst Ring On Time 10 msec Ring Off period in 10 msec units for the fourth cadence on off cycle e Fourth Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the fourth cadence on off cycle SIP User s Manual 370 Document LTRT 65413 SIP User s Manual 8 Auxiliary Configuration Files Note In SIP the Distinctive Ringing pattern is selected according to the Alert Info header in the INVITE message For example Alert Info lt Bellcore dr2 gt or Alert Info lt http Bellcore dr2 gt dr2 defines ringing pattern 2 If the Alert Info header is missing the default ringing tone 0 is played An example of a ringing burst definition is shown below Three ringing bursts followed by repeated ringing of 1 sec on and Bcc cmon NUMBER OF DISTINCTIVE RINGING PATTERNS Number of Ringing Patterns 1 Ringing Pattern 0 Ring Type 0 Freq Hz 25 First Burst Ring On Time 10msec 30 First Burst Ring Off Time 10msec 30 Second Burst Ring On Time 10msec 30 Second Burst Ring Off Time 10mse
626. ues for the supported Classes Of Service refer to Quality of Service Parameters on page 453 10 8 1 1 sn Application 0 OAMP 1 Control 2 Media 3 Media 4 Media 5 Media 6 Media 7 Media 8 Media 9 Media o n 11 Media 12 Media 13 Media 14 Media s a Version 6 0 449 March 2010 7a c tall AudioCodes MediaPack Series 10 8 1 2 Columns of the Multiple Interface Table Each row of the table defines a logical IP interface with its own IP address subnet mask represented by Prefix Length VLAN ID if VLANs are enabled name and application types that are allowed on this interface One of the interfaces may have a default gateway definition Traffic destined to a subnet which does not meet any of the routing rules either local or static routes are forwarded to this gateway as long this application type is allowed on this interface Refer to Gateway Column on page 451 for more details 10 8 1 2 1Index Column This column holds the index of each interface Possible values are 0 to 15 Each interface index must be unique 10 8 1 2 2Application Types Column This column defines the types of applications that are allowed on this interface m OAMP Operations Administration Maintenance and Provisioning applications such as Web Telnet SSH SNMP m CONTROL Call Control Protocols i e SIP m MEDIA RTP streams of Voice m Various combinations of the above mentioned types The following table shows
627. uest This request contains a Supported header with the value gruu The device includes a sip instance Contact header parameter for each contact for which the GRUU is desired This Contact parameter contains a globally unique ID that identifies the device instance The global unique ID is as follows If registration is per endpoint i e the parameter AuthenticationMode is set to 0 it is the MAC address of the device concatenated with the phone number of the endpoint If the registration is per device i e the parameter AuthenticationMode is set to 1 it is only the MAC address When the User Information mechanism is used the globally unique ID is the MAC address concatenated with the phone number of the endpoint defined in the User Info file If the Registrar Proxy supports GRUU the REGISTER responses contain the gruu parameter in each Contact header field The Registrar Proxy provides the same GRUU for the same AOR and instance id in case of sending REGISTER again after expiration of the registration The device places the GRUU in any header field which contains a URI It uses the GRUU in the following messages INVITE requests 2xx responses to INVITE SUBSCRIBE requests 2xx responses to SUBSCRIBE NOTIFY requests REFER requests and 2xx responses to REFER Note If the GRUU contains the opague URI parameter the device obtains the AOR for the user by stripping the parameter The resulting URI is the AOR for exa
628. undant Proxy Registrar server 3 3 4 5 Coders and Profile Definitions The Coders And Profile Definitions submenu includes the following page items m Coders refer to Configuring Coders on page 102 m Coder Group Settings refer to Configuring Coder Groups on page 104 m Tel Profile Settings refer to Configuring Tel Profiles on page 105 m P Profile Settings refer to Configuring IP Profiles on page 107 Implementing the device s Profile features provides the device with high level adaptation when connected to a variety of equipment at both Tel and IP sides and protocols each of which requires different system behavior You can assign different Profiles behavior per call using the call routing tables m Tel to IP Routing page refer to Configuring the Tel to IP Routing on page 126 m IP to Hunt Group Routing Table page refer to Configuring the IP to Hunt Group Routing Table on page 131 In addition you can associate different Profiles per the device s channels Each Profile contains a set of parameters such as coders T 38 Relay Voice and DTMF Gain Silence Suppression Echo Canceler RTP DiffServ Current Disconnect and more The Profiles feature allows you to customize these parameters or turn them on or off per source or destination routing and or per the device s endpoints channels For example specific ports can be assigned a Profile that always uses G 711 Version 6 0 101 March
629. up to 25 indices i e up to 25 different metering rules can be defined This parameter is applicable only to FXS interfaces To associate a charge code to an outgoing Tel to IP call use the Tel to IP Routing To configure the Charge Codes table using the Web interface refer to Configuring the Charge Codes Table on page 113 For an explanation on configuration using ini file table parameters refer to Configuring ini File Table Parameters on page 186 323 March 2010 A c tal AudioCodes MediaPack Series 6 12 Telephone Keypad Sequence Parameters The telephony keypad sequence parameters are described in the table below Table 6 51 Keypad Sequence Parameters Parameter Description Web EMS Call Pickup Key Defines the keying sequence for performing a call pick up Call KeyCallPickup pick up allows the FXS endpoint to answer another telephone s incoming call by pressing this user defined sequence of digits When the user dials these digits e g 77 the incoming call from another phone is forwarded to the user s phone The valid value is a string of up to 15 characters 0 9 and The default is undefined Notes Call pick up is configured only for FXS endpoints pertaining to the same Hunt Group This parameter is applicable only to FXS interfaces Prefix for External Line Prefix2ExtLine Defines a string prefix e g 9 dialed for an external line that when dialed the device plays a
630. uration Parameters Reference Description the SAS agent Each traversed proxy in the path can insert this header causing all future dialogs in the session to pass through it as well When this feature is enabled the SIP Record Route header includes the URI Ir parameter The presence of this parameter indicates loose routing the lack of It indicates strict routing For example Loose routing Record Route lt sip server10 biloxi com Ir gt Strict routing Record Route lt sip bigbox3 site3 atlanta com gt Determines the Proxy Set index number used in SAS Normal mode to forward REGISTER and INVITE requests from the users that are served by the SAS application The valid range is 0 to 5 The default value is 0 i e default Proxy Set Determines the Proxy Set index number used in SAS Emergency mode for fallback when the user is not found in the Registered Users database Each time a new SIP request arrives the SAS application checks whether the user is listed in the registration database If the user is located in the database the request is sent to the user If the user is not found the request is forwarded to the next redundant SAS defined in the Redundant SAS Proxy Set If that SAS Proxy IP appears in the Via header of the request it is not forwarded thereby preventing loops in the request s course If no such redundant SAS exists the SAS sends the request to its default gateway configured by the parameter SAS
631. urst Action Upon Match AccessList Allow Type Match Count AccessList MatchCount SIP User s Manual MediaPack Series Description A read only field indicating whether the rule is active or not Note After device reset all rules are active IP address or DNS name of source network or a specific host IP network mask 32 for a single host or the appropriate value for the source IP addresses A value of 8 corresponds to IPv4 subnet class A network mask of 255 0 0 0 A value of 16 corresponds to IPv4 subnet class B network mask of 255 255 0 0 A value of 24 corresponds to IPv4 subnet class C network mask of 255 255 255 0 The IP address of the sender of the incoming packet is trimmed in accordance with the prefix length in bits and then compared to the parameter Source IP The destination UDP TCP ports on this device to which packets are sent The valid range is 0 to 65535 Note When the protocol type isn t TCP or UDP the entire range must be provided The protocol type e g UDP TCP ICMP ESP or Any or the IANA protocol number in the range of 0 Any to 255 Note This field also accepts the abbreviated strings SIP and HTTP Specifying these strings implies selection of the TCP or UDP protocols and the appropriate port numbers as defined on the device Maximum allowed packet size The valid range is 0 to 65535 Note When filtering fragmented IP packets this fiel
632. using the Web interface refer to Configuring the NFS Settings on page 56 For a description of configuring ini file table parameters refer to Configuring ini File Table Parameters on page 186 217 March 2010 ca AudioCodes 6 1 7 MediaPack Series DNS Parameters The Domain name System DNS parameters are described in the table below Parameter Web DNS Primary Server IP EMS DNS Primary Server DNSPriServerIP Web DNS Secondary Server IP EMS DNS Secondary Server DNSSecServerlP Web Internal DNS Table EMS DNS Information DNS2IP Web Internal SRV Table EMS DNS Information SRV2IP SIP User s Manual Table 6 7 DNS Parameters Description The IP address of the primary DNS server Enter the IP address in dotted decimal notation for example 10 8 2 255 Notes For this parameter to take effect a device reset is required To use Fully Qualified Domain Names FQDN in the Tel to IP Routing you must define this parameter The IP address of the second DNS server Enter the IP address in dotted decimal notation for example 10 8 2 255 Note For this parameter to take effect a device reset is required This ini file table parameter configures the internal DNS table for resolving host names into IP addresses Up to four different IP addresses in dotted decimal notation can be assigned to a host name The format of this parameter is as follows Dns2lp FORMAT Dns2Ip_
633. uthorization and Accounting AAA indications 0 None No indications default 3 Accounting Only Only accounting indications are used Defines the device s response upon a RADIUS timeout 0 Deny Access Denies access 1 Verify Access Locally Checks password locally default Number of concurrent calls that can communicate with the RADIUS server optional The valid range is 0 to 240 The default value is 240 Number of retransmission retries The valid range is 1 to 10 The default value is 3 240 Document LTRT 65413 SIP User s Manual Parameter RadiusTO Web RADIUS Authentication Server IP Address RADIUSAuthServerlP RADIUSAuthPort Web RADIUS Shared Secret SharedSecret Web Default Access Level DefaultAccessLevel Web Local RADIUS Password Cache Mode RadiusLocalCacheMode Web Local RADIUS Password Cache Timeout RadiusLocalCacheTimeout Web RADIUS VSA Vendor ID RadiusVSAVendorlD Web RADIUS VSA Access Level Attribute RadiusVSAAccessAttribute Version 6 0 6 Configuration Parameters Reference Description Determines the time interval measured in seconds the device waits for a response before a RADIUS retransmission is issued The valid range is 1 to 30 The default value is 10 IP address of the RADIUS authentication server Note For this parameter to take effect a device reset is required RADIUS Authentication Server Port Note For this
634. uting refer to Configuring the IP to Hunt Group Routing Table on page 131 m Internal DNS Table refer to Configuring the Internal DNS Table on page 134 Internal SRV Table refer to Configuring the Internal SRV Table on page 134 m Forward on Busy Trunk Dest refer to Configuring Call Forward upon Busy Trunk on page 135 Configuring Reasons for Alternative Routing The Reasons for Alternative Routing page allows you to define up to four different call release termination reasons for IP to Tel call releases and for Tel to IP call releases If a call is released as a result of one of these reasons the device tries to find an alternative route for that call The device supports up to two different alternative routes The release reasons depends on the call direction m Release reason for IP to Tel calls provided in Q 931 notation As a result of a release reason an alternative Hunt Group is provided For defining an alternative Hunt Group refer to Configuring the IP to Hunt Group Routing Table on page 131 This call release reason type can be configured for example when the destination is busy and release reason 17 is issued or for other call releases that issue the default release reason 3 refer to the parameter DefaultReleaseCause m Release reason for Tel to IP calls provided in SIP 4xx 5xx and 6xx response codes As a result of a release reason an alternative IP address is provided For defining an altern
635. utomatic Dialing on page 137 Caller Display Information refer to Configuring Caller Display Information on page 138 Call Forward refer to Configuring Call Forward on page 140 Caller ID Permissions refer to Configuring Caller ID Permissions on page 141 Call Waiting refer to Configuring Call Waiting on page 142 3 3 4 9 1 Configuring Authentication The Authentication page defines a user name and password for authenticating each device port Authentication is typically used for FXS interfaces but can also be used for FXO interfaces For configuring whether authentication is performed per port or for the entire device use the parameter AuthenticationMode If authentication is for the entire device the configuration on this page is ignored If either the user name or password fields are omitted the port s phone number and global password using the Password parameter are used instead After you click the Submit button the password is displayed as an asterisk You can also configure Authentication using the ini file table parameter Authentication refer to SIP Configuration Parameters on page 245 SIP User s Manual 136 Document LTRT 65413 SIP User s Manual 3 Web Based Management gt To configure the Authentication Table 1 Set the parameter Authentication Mode AuthenticationMode to Per Endpoint 2 Open the Authentication page Configuration tab gt Protocol Configura
636. utton to start the software upgrad Warning Once software update commences the upgrade process cannot be cancelled In case of upgrade failure the device will reset and the previous configuration burned to flash will be restored 3 Click the Start Software Upgrade button the Load a CMP file Wizard page appears Note At this stage you can quit the Software Update Wizard by clicking Cancel x without requiring a device reset However once you start uploading a cmp file the process must be completed with a device reset Version 6 0 169 March 2010 7a e AudioCodes MediaPack Series 4 Click the Browse button navigate to the cmp file and then click Send File the cmp file is loaded to the device and you re notified as to a successful loading 5 Click one of the following buttons Y Reset the device resets with the newly loaded cmp utilizing the existing configuration and auxiliary files gt e Next the Load an ini File wizard page opens Note that as you progress by clicking Next the relevant file name corresponding to the applicable Wizard page is highlighted in the file list on the left 6 Inthe Load an ini File page you can now choose to either e Click Browse navigate to the ini file and then click Send File the ini file is loaded to the device and you re notified as to a successful loading e Use the ini file currently used by the device by not selecting an ini file and
637. v Outgoing calls are tagged according to this parameter Defines the DiffServ value for the Gold CoS content The valid range is 0 to 63 The default value is 26 Defines the DiffServ value for the Bronze CoS content The valid range is 0 to 63 The default value is 10 The Network Address Translation NAT and Simple Traversal of UDP through NAT STUN parameters are described in the table below Table 6 5 NAT and STUN Parameters Parameter STUN Parameters Web Enable STUN EMS STUN Enable EnableSTUN Description Determines whether Simple Traversal of UDP through NATs STUN is enabled 0 Disable default 1 Enable When enabled the device functions as a STUN client and communicates with a STUN server located in the public Internet STUN is used to discover whether the device is located behind a NAT and the type of NAT In addition it is used to determine the IP addresses and port numbers that the NAT assigns to outgoing signaling messages using SIP and media streams using RTP RTCP and T 38 STUN works with many existing NAT types and does not require any special behavior from them For detailed information on STUN refer to STUN on page 444 Notes For this parameter to take effect a device reset is required For defining the STUN server domain name use the parameter STUNServerDomainName SIP User s Manual 214 Document LTRT 65413 SIP User s Manual Parameter Web
638. v for incoming MLPP calls with the Resource Priority header The valid range is 0 to 63 The default value is 50 Notes The same value must be configured for this parameter and the parameter PremiumServiceClassControlDiffServ Outgoing calls are tagged according to the parameter PremiumServiceClassControlDiffServ Defines the E911 or Emergency Telecommunication Services ETS MLPP Preemption mode 0 Standard Mode ETS calls have the highest priority and preempt any MLPP call default 1 Treat as routine mode ETS calls are handled as routine calls Defines the index of the Precedence Ringing tone in the Call Progress Tones CPT file This tone is used when the parameter CallPriorityMode is set to 1 and a Precedence call is received from the IP side The valid range is 1 to 16 The default value is 1 i e plays standard Ringing tone 308 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Multiple Differentiated Services Code Points DSCP per MLPP Call Priority Level Precedence Parameters The MLPP service allows placement of priority calls where properly validated users can preempt terminate lower priority phone calls with higher priority calls For each MLPP call priority level the DSCP can be set to a value from 0 to 63 The Resource Priority value in the Resource Priority SIP header can be one of the following MLPP Precedence Level Preced
639. ve Blind Transfer REFER Blind transfer is performed after a call is established between A and B and party A decides to immediately transfer the call to C without speaking with C The result of the transfer is a call between B and C similar to consultation transfer but skipping the consultation stage Call transfer is initiated by sending REFER with REPLACES The device can receive and act upon receiving REFER with or without REPLACES The device can receive and act upon receiving INVITE with REPLACES in which case the old call is replaced by the new one The INVITE with REPLACES can be used to implement Directed Call Pickup SIP User s Manual 412 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 7 5 Call Forward The following methods of call forwarding are supported Immediate incoming call is forwarded immediately and unconditionally Busy incoming call is forwarded if the endpoint is busy No Reply incoming call is forwarded if it isn t answered for a specified time On Busy or No Reply incoming call is forwarded if the port is busy or when calls are not answered after a specified time Do Not Disturb immediately reject incoming calls Upon receiving a call for a Do Not Disturb the 603 Decline SIP response code is sent Three forms of forwarding parties are available Served party party configured to forward the call FXS device Originating party party that initiates the first c
640. ve VLAN 0 Sends priority tag packets default 1 Sends regular packets with no VLAN tag Note For this parameter to take effect a device reset is required 6 1 3 Static Routing Parameters The static routing parameters are described in the table below Table 6 3 Static Routing Parameters Parameter Description Static IP Routing Table Parameters You can define up to 50 static IP routing rules for the device For example you can define static routing rules for the OAMP and Control networks since a default gateway is supported only for the Media traffic network Before sending an IP packet the device searches this table for an entry that matches the requested destination host network If such an entry is found the device sends the packet to the indicated router If no explicit entry is found the packet is sent to the default gateway configured in the Multiple Interface table The IP routing parameters are array parameters Each parameter configures a specific column in the IP Routing table The first entry in each parameter refers to the first row in the IP Routing table the second entry to the second row and so on In the following example two rows are configured when the device is in network 10 31 x x RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 255 255 0 RoutingTableGatewaysColumn 10 31 0 1 10 31 0 112 RoutingTablelnterfacesColumn
641. voice stream 1 Relay Currently not applicable 3 Mute The caller ID signal is detected from the Tel side and then erased from the voice stream default Note Caller ID detection is applicable only to FXO interfaces 6 8 2 Call Waiting Parameters The call waiting parameters are described in the table below Table 6 35 Call Waiting Parameters Parameter Web EMS Enable Call Waiting EnableCallWaiting EMS Send 180 For Call Waiting Send180ForCallWaiting SIP User s Manual Description Determines whether Call Waiting is enabled 0 Disable Disable the Call Waiting service 1 Enable Enable the Call Waiting service default If enabled when an FXS interface receives a call on a busy endpoint it responds with a 182 response and not with a 486 busy The device plays a call waiting indication signal When hook flash is detected the device switches to the waiting call The device that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received Notes The device s Call Progress Tones CPT file must include a Call Waiting Ringback tone caller side and a Call Waiting tone called side FXS only The EnableHold parameter must be enabled on both the calling and the called side You can use the ini file table parameter CallWaitingPerPort to enable Call Waiting per port For information on the Call Waiting feature refer t
642. w some examples of selected network configurations and their matching ini file configuration Example 1 Single Interface Configuration Multiple Interface table with a single interface for OAMP Media and Control applications Table 10 9 Multiple Interface Table Example1 Index Application Interface IP Address eh Pte sus u maraca Length Gateway ID Name OAMP 0 Media amp IPv4 192 168 85 14 16 192 168 0 1 1 mylnterface Control VLANS are not required and the Native VLAN ID is irrelevant Class of Service parameters may have default values The required routing table features two routes Table 10 10 Routing Table Example 1 Destination Prefix Length Subnet Mask Gateway Interface Metric 201 201 0 0 16 192 168 0 2 0 1 202 202 0 0 16 192 168 0 3 0 1 The DNS NTP applications may have their default application types This example s matching ini file is shown above However since many parameter values equal their default values they can be omitted The ini file can be also written as follows Interface Table Configuration InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes InterfaceTable InterfaceMode InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway InterfaceTable VlanID InterfaceTable InterfaceName mtenrtaceTablle 6 10 194 168 8514 16 194 168 01 I m2 InterfaceTable Routing Table Configuration RoutingTableDestinationsColumn 201 201 0
643. way Interface Name o O MI lv 4 VLAN Mode Disable Native VLAN ID h 4 In the Add Index field enter the desired index number for the new interface and then click Add the index row is added to the table 5 Configure the interface according to the table below 6 Click the Apply button the interface is added to the table and the Done button appears 7 Click Done to validate the interface If the interface is not a valid e g if it overlaps with another interface in the table or it does not adhere to the other rules for adding interfaces a message is displayed to inform you and you must redefine your interfaces accordingly 8 To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 4 Multiple Interface Table Parameters Description Parameter Description Table parameters Index Index of each interface The range is 0 to 15 Note Each interface index must be unique Web Application Type Types of applications that are allowed on the specific EMS Application Types interface InterfaceTable_ApplicationTypes 0 OAMP Only Operations Administration Maintenance and Provisioning OAMP applications e g Web Telnet SSH and SNMP are allowed on the interface 1 Media Only Media i e RTP streams of voice is allowed on the interface 2 Control Only Call Control applications e g SIP are allowed on the interface 3 OAMP Media Only
644. without SDP is received If 180 183 with SDP message is received the device cuts through the voice channel and doesn t play RBT default 3 Play Local Until Remote Media Arrive Plays the RBT according to received media The behaviour is similar to 2 If a SIP 180 response is received and the voice channel is already open due to a previous 183 early media response or due to an SDP in the current 180 response the device plays a local RBT if there are no prior received RTP packets The device stops playing the local RBT as soon as it starts receiving RTP packets At this stage if the device receives additional 18x responses it does not resume playing the local RBT 320 Document LTRT 65413 SIP User s Manual 6 Configuration Parameters Reference Parameter Description Web Play Ringback Tone to IP Determines whether or not the device plays a ringback tone RBT EMS Play Ring Back Tone To to the IP side for IP to Tel calls IP PlayRBTone2IP 0 Don t Play Ringback tone isn t played default 1 Play Ringback tone is played after SIP 183 session progress response is sent Notes This parameter is applicable only to FXS interfaces To enable the device to send a 183 180 SDP responses set the parameter EnableEarlyMedia to 1 If the parameter EnableDigitDelivery is set to 1 the device doesn t play a ringback tone to IP and doesn t send 183 or 180 SDP responses 6 11 2 Tone Detection Parameter
645. x AddPhoneContextAsPrefix NPI TON Phone Context Version 6 0 Description Determines whether the received Phone Context parameter is added as a prefix to the outgoing Called and Calling numbers 0 Disable Disable default 1 Enable Enable Select the Number Plan assigned to this entry 0 Unknown Unknown default 1 E 164 Public E 164 Public 9 Private Private Select the Type of Number assigned to this entry If you selected Unknown as the NPI you can select Unknown 0 If you selected Private as the NPI you can select one of the following v 0 Unknown v 1 Level 2 Regional v 2 Level 1 Regional v B PSTN Specific v 4 Level 0 Regional Local If you selected E 164 Public as the NPI you can select one of the following 0 Unknown 1 International 2 National 3 Network Specific 4 Subscriber 6 Abbreviated The Phone Context SIP URI parameter ANARAN 123 March 2010 7a c tall AudioCodes MediaPack Series 3 3 4 8 3 3 4 8 1 Routing Tables The Routing Tables submenu allows you to configure call routing rules This submenu includes the following page items m Alternative Routing refer to Configuring Reasons for Alternative Routing on page 124 m Routing General Parameters refer to Configuring Routing General Parameters on page 125 Tel to IP Routing refer to Configuring the Tel to IP Routing on page 126 m P to Trunk Group Ro
646. x and 6xx response codes The format of this parameter is as follows AltRouteCauseTel2IP FORMAT AltRouteCauseTel2IP Index AltRouteCauseTel2IP ReleaseCause AltRouteCauseTel2IP For example AltRouteCauseTel2IP 0 486 Busy Here AltRouteCauseTel2IP 1 480 Temporarily Unavailable AltRouteCauseTel2IP 2 408 No Response Notes This parameter can include up to 5 indices The reasons for alternative routing for Tel to IP calls apply only when a Proxy is not used When there is no response to an INVITE message after INVITE retransmissions the device issues an internal 408 No Response implicit release reason The device sends the call to an alternative IP route only after the call has failed and the device has subsequently attempted twice to establish the call unsuccessfully The device also plays a tone to the endpoint whenever an alternative route is used This tone is played for a user defined time configured by the parameter AltRoutingToneDuration 339 March 2010 Aa tal AudioCodes MediaPack Series Parameter Description Foran explanation on using ini file table parameters refer to Configuring ini File Table Parameters on page 186 Web Reasons for Alternative IP to Tel Routing Table EMS Alt Route Cause IP to Tel AltRouteCauselP2Tel This ini file table parameter configures call failure reason values received from the Tel side If a call is released as a result of one
647. x refer to Number Manipulation and Routing Parameters on page 331 gt To configure IP to Tel routing rules 1 Open the IP to Hunt Group Routing Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt IP to Trunk Group Routing page item Figure 3 80 Inbound IP Routing Table Page Routing Index 1 12 v IP To Tel Routing Mode Route calls before manipulation W Hunt Dest Host Prefix Source Host Prefix Dest Phone Prefix Source Phone Prefix Source IP Address gt Group 7 ID 501 502 Jim IP Profile Source ID IPGroup ID 2 1 ii 2 F omancon p p J ik L A J 1013 645 B The figure above shows the following configured IP to Tel routing rules e Rule 1 If the incoming IP call destination phone prefix is between 10 and 19 the call is assigned settings configured for IP Profile ID 2 and routed to Hunt Group ID 1 Version 6 0 131 March 2010 A L e AudioCodes MediaPack Series e Rule 2 If the incoming IP call destination phone prefix is between 501 and 502 and source phone prefix is 101 the call is assigned settings configured for IP Profile ID 1 and routed to Hunt Group ID 2 e Rule 3 If the incoming IP call has a From URI host prefix as domain com the call is routed to Hunt Group ID 3 2 From the Routing Index drop down list select the range of entries that you want to add 3 Configure
648. x corresponding to the table row that you want to edit 2 Click Edit the fields in the corresponding index row become available 3 Modify the values as required and then click Apply the new settings are applied gt To organize the index entries in ascending consecutive order m Click Compact the index entries are organized in ascending consecutive order starting from index 0 For example if you added three index entries 0 4 and 6 then the index entry 4 is re assigned index number 1 and the index entry 6 is re assigned index number 2 Figure 3 12 Compacting a Web Interface Table VlanID InterfaceName Duplicate Compact Index ApplicattonTypes IPv6InterfaceMode IPAddress PrefixLength Gateway VianIO InterfaceName O 1 O gt To delete an existing index table entry 1 In the Index column select the index corresponding to the table row that you want to delete 2 Click Delete the table row is removed from the table Version 6 0 35 March 2010 7a c tal AudioCodes MediaPack Series 3 1 7 Searching for Configuration Parameters The Web interface provides a search engine that allows you to search any ini file parameter that is configurable by the Web interface i e has a corresponding Web parameter You can search for a specific parameter e g EnablelPSec or a sub string of that parameter e g sec If you search for a sub string all parameters that contain the searched sub s
649. xample 97120155 e Index 2 When the source number has prefix 1001 e g 1001876 it is changed to 587623 e Index 3 When the source number has prefix 123451001 e g 1234510012001 it is changed to 20018 Version 6 0 117 March 2010 A L tal AudioCodes MediaPack Series e Index 4 When the source number has prefix from 30 to 40 and a digit e g 3122 it is changed to 2312 e Index 5 When the destination number has the prefix 6 7 or 8 e g 85262146 source number prefix 2001 it is changed to 3146 aE 7 From the Table Index drop down list select the range of entries that you want to edit Configure the Number Manipulation table according to the table below Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 18 Number Manipulation Parameters Description Parameter Source Trunk Group Source IP Group Web Destination Prefix EMS Prefix Web EMS Source Prefix Web EMS Source IP SIP User s Manual Description The source Hunt Group ID for Tel to IP calls To denote any Hunt Group leave this field empty Notes The value 1 indicates that it is ignored in the rule This parameter is available only in the Source Phone Number Manipulation Table for Tel gt IP Calls and Destination Phone Number Manipulation Table for Tel gt IP Calls pages For IP to IP call routing this parameter
650. xist or if the phone number isn t found the device sends a SIP response to IP according to the parameter DefaultReleaseCause This parameter defines Q 931 release causes Its default value is 3 which is mapped to the SIP 404 response By changing its value to 34 the SIP 503 response is sent Other causes can be used as well Querying Device Channel Resources using SIP OPTIONS The device reports its maximum and available channel resources in SIP 200 OK responses upon receipt of SIP OPTIONS messages The device sends this information in the SIP X Resources header with the following parameters m telchs specifies the total telephone channels as well as the number of free available telephone channels m mediachs not applicable Below is an example of the X Resources X Resources telchs 8 4 mediachs 0 0 In the example above telchs specifies the number of available channels and the number of occupied channels 4 channels are occupied and 8 channels are available SIP User s Manual 432 Document LTRT 65413 SIP User s Manual 9 IP Telephony Capabilities 9 11 Event Notification using X Detect Header The device supports the sending of notifications to a remote party notifying the occurrence or detection of certain events on the media stream Event detection and notifications is performed using the SIP X Detect message header and only when establishing a SIP dialog For supporting some events
651. y supports only the T 38 UDP syntax T 38 can be configured in the following ways m Switching to T 38 mode using SIP Re INVITE messages refer to Switching to T 38 Mode using SIP Re INVITE on page 401 m Automatically switching to T 38 mode without using SIP Re INVITE messages refer to Automatically Switching to T 38 Mode without SIP Re INVITE on page 402 When fax transmission ends the reverse switching from fax relay to voice is automatically performed at both the local and remote endpoints You can change the fax rate declared in the SDP using the parameter FaxRelayMaxRate this parameter doesn t affect the actual transmission rate In addition you can enable or disable Error Correction Mode ECM fax mode using the FaxRelayECMEnable parameter When using T 38 mode you can define a redundancy feature to improve fax transmission over congested IP networks This feature is activated using the FaxRelayRedundancyDepth and FaxRelayEnhancedRedundancyDepth parameters Although this is a proprietary redundancy scheme it should not create problems when working with other T 38 decoders 9 6 2 1 1 Switching to T 38 Mode using SIP Re INVITE In the Switching to T 38 Mode using SIP Re INVITE mode upon detection of a fax signal the terminating device negotiates T 38 capabilities using a Re INVITE message If the far end device doesn t support T 38 the fax fails In this mode the parameter FaxTransportMode is ignored To configure T 3
652. y ID RPI headers for calling and called numbers for Tel to IP calls Three Way Conferencing The device supports three way conference calls These conference calls can also occur simultaneously The following example demonstrates three way conferencing This example assumes that a telephone A connected to the device wants to establish a three way conference call with two remote IP phones B and C 1 User A has an ongoing call with IP phone B 2 User A places IP phone B on hold by pressing the telephone s flash hook button defined by the parameter HookFlashCode 3 User A hears a dial tone and then makes a call to IP phone C 4 IP phone C answers the call 5 User A can now establish a three way conference call between A B and C by pressing the flash hook button defined by the parameter ConferenceCode e g regular flash hook button or 1 Instead of using the flash hook button to establish a three way conference call you can dial a user defined hook flash code e g 1 configured by the parameter HookFlashCode Three way conferencing is applicable only to FXS interfaces Version 6 0 419 March 2010 7a i L tal AudioCodes MediaPack Series The device supports the following conference modes configured by the parameter 3WayConferenceMode Local on board conferencing whereby the conference is established on the device without the need for an external Conference server This feature includes local m
653. you need to enable the keep alive with Proxy option by setting the parameter EnableProxyKeepAlive to 1 or 2 Defines the time interval in seconds between each Proxy IP list refresh The range is 5 to 2 000 000 The default interval is 60 Determines whether the device falls back to the Tel to IP Routing for call routing when Proxy servers are unavailable 0 Disable Fallback is not used default 1 Enable The Tel to IP Routing is used when Proxy servers are unavailable When the device falls back to the Tel to IP Routing it continues scanning for a Proxy When the device locates an active Proxy it switches from internal routing back to Proxy routing Note To enable the redundant Proxies mechanism set the parameter EnableProxyKeepAlive to 1 or 2 Determines whether the device s internal routing table takes precedence over a Proxy for routing calls 0 No Only a Proxy server is used to route calls default 1 Yes The device checks the routing rules in the Tel to IP Routing for a match with the Tel to IP call Only if a match is not found is a Proxy used Determines whether the device sends SIP messages and responses through a Proxy server 0 Disable Use standard SIP routing rules default 1 Enable All SIP messages and responses are sent to the Proxy server Note This parameter is applicable only if a Proxy server is used i e the parameter IsProxyUsed is set to 1
654. you to configure up to 120 SAS routing rules for Normal and Emergency modes The device routes the SAS call received SIP INVITE message once a rule in this table is matched If the characteristics of an incoming call do not match the first rule the call characteristics is then compared to the settings of the second rule and so on until a matching rule is located If no rule is matched the call is rejected When SAS receives a SIP INVITE request from a proxy server the following routing logic is performed a Sends the request according to rules configured in the IP2IP Routing table b If no matching routing rule exists the device sends the request according to its SAS registration database c If no routing rule is located in the database the device sends the request according to the Request URI header Note The IP2IP Routing table can also be configured using the ini file table parameter IP2IPRouting refer to SIP Configuration Parameters on page 245 gt To configure the IP2IP Routing table for SAS 1 In the SAS Configuration page refer to Configuring Stand Alone Survivability Parameters on page 145 click the SAS Routing Table gt button the IP2IP Routing Table page appears 2 Add an entry and then configure it according to the table below 3 Click the Apply button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 161 Table 3 25 SAS Routing T
655. ypad sequence that activates the delayed hotline option To activate the delayed hotline option from the telephone perform the following 1 Dial the user defined sequence number on the keypad a dial tone is heard 2 Dial the telephone number to which the phone automatically dials after a configurable delay terminate the number with a confirmation tone is heard Keypad sequence that deactivates the delayed hotline option After the sequence is pressed a confirmation tone is heard Keypad Feature Transfer Parameters Web Blind EMS Blind Transfer KeyBlindTransfer SIP User s Manual Keypad sequence that activates blind transfer for Tel to IP calls There are two possible scenarios Option 1 After this sequence is dialed the current call is put on hold using Re INVITE a dial tone is played to the phone and then phone number collection starts Option 2 A Hook Flash is pressed the current call is put on hold a dial tone is played to the phone and then digit collection starts After this sequence is identified the device continues the collection of the destination phone number For both options after the phone number is collected it s sent to the transferee in a SIP REFER request without a Replaces header The call is then terminated and a confirmation tone is played to the phone If the phone number collection fails due to a mismatch a reorder tone is played to the phone Notes This parameter is
656. ype AnalogSignalTransportType Web RTP Redundancy Depth EMS Redundancy Depth RTPRedundancyDepth Version 6 0 Description Minimum delay in msec for the Dynamic Jitter Buffer The valid range is 0 to 150 The default delay is 10 Note For more information on Jitter Buffer refer to Dynamic Jitter Buffer Operation on page 441 Dynamic Jitter Buffer frame error delay optimization factor The valid range is 0 to 13 The default factor is 10 Notes For data fax and modem calls set this parameter to 13 For more information on Jitter Buffer refer to Dynamic Jitter Buffer Operation on page 441 Determines the analog signal transport type 0 Ignore Analog Signals Ignore default 1 RFC 2833 Analog Signal Relay Transfer hookflash using RFC 2833 Determines whether the device generates redundant packets This can be used for packet loss where the missing information audio can be reconstructed at the receiver end from the redundant data that arrives in the subsequent packet s 0 0 Disable the generation of redundant packets default 1 1 Enable the generation of RFC 2198 redundancy packets payload type defined by the parameter RFC2198PayloadType Note The RTP redundancy dynamic payload type can be included in the SDP by using the parameter EnableRTPRedundancyNegotiation 357 March 2010 ca AudioCodes Parameter Web Enable RTP Redundancy Negotiation EnableRTP

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