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User`s Manual Version 5.0

1. 2 From the Routing Index drop down list select the range of entries that you want to edit up to 24 entries can be configured 3 In the Hunt Group ID field enter the hunt group ID number 4 From the Channel Select Mode drop down list select the Channel Select Mode that determines the method in which new calls are assigned to channels within the hunt groups entered in the field to the right of this field For information on available Channel Select Modes refer to Table 5 26 on page 118 5 From the Registration Mode drop down list select the registration mode 6 Repeat steps 4 and 5 for each defined hunt group 7 Click the Submit button to save your changes 8 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Version 5 0 117 December 2006 Ta WH wt AudioCodes MediaPack Table 5 26 Channel Select Modes Mode Description By Dest Phone Number Select the gateway port according to the called number refer to the note below Cyclic Ascending Select the next available channel in ascending cycle order Always select the next higher channel number in the hunt group When the gateway reaches the highest channel number in the hunt group it selects the lowest channel number in the hunt group and then starts ascending again Ascending Select the lowest available channel Always
2. 3 Configure the following settings for both gateways In the Tel to IP Routing screen in the first row enter 10 in the Destination Phone Prefix field and enter the IP address of the first gateway 10 2 37 10 in the field IP Address In the second row enter 20 and the IP address of the second gateway 10 2 37 20 respectively These settings enable the routing from both gateways of outgoing Tel gt IP calls that start with 10 to the first gateway and calls that start with 20 to the second gateway Dest Phone Prefix Source Phone Prefix Dest IP Address Profile ID Status 1 10 10 2 37 10 0 2 eo o F 10 2372 fo 4 Make a call Pick up the phone connected to port 1 of the first MediaPack and dial 102 to the phone connected to port 2 of the same gateway Listen out for progress tones at the calling endpoint and for ringing tone at the called endpoint Answer the called endpoint speak into the calling endpoint and check the voice quality Dial 201 from the phone connected to port 1 of the first MediaPack gateway the phone connected to port 1 of the second MediaPack rings Answer the call and check the voice quality Version 5 0 237 December 2006 Ta laS AudioCodes 8 12 2 SIP Call Flow MediaPack The following Call Flow describes SIP messages exchanged between two MediaPack gateways during simple call Telephone 6000 dials 2000
3. Reader s Notes SIP User s Manual 208 Document LT RT 65408 SIP User s Manual 6 ini File Configuration of the MediaPack 6 ini File Configuration of the MediaPack As an alternative to configuring the VoIP gateway using the Web Interface refer to Chapter 5 on page 49 it can be configured by loading the ini file containing Customer configured parameters The ini file is loaded via the BootP TFTP utility refer to Appendix C on page 349 or via any standard TFTP server It can also be loaded through the Web Interface refer to Section 5 6 3 on page 165 The ini file configuration parameters are stored in the MediaPack non volatile memory after the file is loaded When a parameter is missing from the ini file a default value is assigned to that parameter according to the cmp file loaded on the MediaPack and stored in the non volatile memory thereby overriding the value previously defined for that parameter Therefore to restore the default configuration parameters use the ini file without any valid parameters or with a semicolon preceding all lines in the file Some of the MediaPack parameters are configurable through the ini file only and not via the Web These parameters usually determine a low level functionality and are seldom changed for a specific application Note For detailed explanation of each parameter refer to Chapter 5 on page 49 6 1 Secured ini File The ini file contains sensiti
4. Version 5 0 247 December 2006 Ta WH wt AudioCodes MediaPack 9 2 1 The way SIP is designed creates a problem for VoIP traffic to pass through NAT SIP uses IP addresses and port numbers in its message body 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 gateway located behind a NAT that initiates a signaling path will have problems in receiving incoming signaling responses they will be blocked by the NAT Furthermore the initiating gateway must notify the receiving gateway where to send the media to To solve these problems the following mechanisms are available m STUN refer to Section 9 2 1 E First Incoming Packet Mechanism refer to Section 9 2 2 on page 249 E RTP No Op packets according to the avt rtp noop draft refer to Section 9 2 3 on page 249 E For SNMP NAT traversal refer to Section 14 10 on page 322 STUN Simple Traversal of UDP through NATs STUN according to 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 signaling and the media streams STUN works with many existing NAT types and does not require any special behavior from them STUN enables the gateway to discover the presence and types
5. 2 From the Table Index drop down list select the range of entries that you want to edit up to 20 entries can be configured for Source Number Manipulation and 50 entries for Destination Number Manipulation 3 Configure the Number Manipulation table according to Table 5 13 Version 5 0 91 December 2006 Ta WH wi AudioCodes MediaPack 4 Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 13 Number Manipulation Parameters continues on pages 92 to 93 Parameter Description Dest Prefix Each entry in the Destination Prefix fields represents a destination telephone number prefix An asterisk represents any number Source Prefix Each entry in the Source Prefix fields represents a source telephone number prefix An asterisk represents any number Source IP Each entry in the Source IP fields represents the source IP address of the call obtained from the Contact header in the INVITE message This column only applies to the Destination Phone Number Manipulation Table for IP to Tel Note The source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The manipulation rules are applied to any incoming call whose Destination number prefix matches the prefix defined in the Destination Number
6. Device Behavior Upon RADIUS Defines the gateway s operation if a response isn t received from the Timeout RADIUS server after the 5 seconds timeout expires BehaviorUponRadiusTimeout Deny Access 0 the gateway denies access to the Web and Telnet embedded servers Verify Access Locally 1 the gateway checks the local username and password default SIP User s Manual 174 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 50 Security Settings General Security Settings Parameters Parameter Local RADIUS Password Cache Mode RadiusLocalCacheMode Local RADIUS Password Cache Timeout RadiusLocalCacheTimeout RADIUS VSA Vendor ID RadiusVSAVendorID RADIUS VSA Access Level Attribute RadiusVSAAccessAttribute EtherDiscover Settings EtherDiscover Operation Mode SRTP Seitings Enable Media Security EnableMediaSecurity Media Security Behavior MediaSecurityBehaviour IPSec Settings Enable IP Security EnablelPSec continues on pages 174 to 175 Description Defines the gateway s mode of operation regarding the timer configured by the parameter RadiusLocalCacheTimeout that determines the validity of the username and password verified by the RADIUS server Absolute Expiry Timer 0 when you access a Web screen the timeout doesn t reset but rather continues decreasing Reset Timer Upon Access 1 upon each access to a Web screen the timeout always r
7. Version 5 0 189 December 2006 Ta WH wt AudioCodes MediaPack Table 5 59 Call Counters Description continues on pages 189 to 190 Counter Number of Failed Calls due to a Busy Line Number of Failed Calls due to No Answer Number of Failed Calls due to No Route Number of Failed Calls due to No Matched Capabilities Number of Failed Calls due to Other Failures Percentage of Successful Calls Average Call Duration sec Attempted Fax Calls Counter Successful Fax Calls Counter Description This counter 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 This counter indicates the number of calls that weren t answered It is incremented as a result of one of the following release reasons GWAPP_NO_USER_RESPONDING 18 GWAPP_NO_ANSWER_FROM_USER_ALERTED 19 And when the call duration is zero as a result of the following GWAPP_NORMAL_CALL_CLEAR 16 This counter 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 This counter indicates the number of calls that failed due to mismatched gateway 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 param
8. 8 6 8 7 8 8 8 9 8 10 St 8 11 8 12 Ci SIP User s Manual 6 Document LTRT 65408 SIP User s Manual Contents 4 Me Veb Server 2 9 Networking Capabil Ae aeliee ans es ie 9 1 Eihera Interface Cc 10 11 Special Applications Metering PORES Relay cnica Version 5 0 7 December 2006 7a wt AudioCodes MediaPack T Sen a a 12 1 IP ec ma E E E A A EE ear net A A T E E 14 5 Total Cour 1 ters A aerneeciaemnaneotn SIP User s Manual l 8 Document LTRT 65408 SIP User s Manual Contents ap M juring Trap M uring Pesi Man C 1 Version 5 0 9 l December 2006 7a wt AudioCodes MediaPack C 9 Log PON i SE REO eC ee EL eerie ART RETRE EE EE T tert emer ee OE 6 l ai d Lir C 12 ae Client 7 Templates AEAEE EA E A D RTP RTCP Payload Types and Port Allocation csseeseseneeneenen D 1 Packet Types Defined in RFC 3551 D 2 Defined Payload Types D 3 Default RTP RTCPIT 38 Port Allo E aA i and Tools F SNM P Traps nannnnnnnnnnn EEE I EE EEA E E AA EE PA A E TE A A OD F 1 Alarm Traps PIE E NE T E EERE F 1 1 ye F 1 2 F 1 3 FiA Trap Verbin G Installation and Configuration of Apache HTTP Server sessseseees 383 G 1 Windows 2000 XP Operation Systems cccseeeceeeeeeeeeeeeneeees pectin ee G2 Linu Operation Sete ercan a ania H Regulatory TFIID a iceiceecnirininisiccircasasasavn
9. Note that when remote configuration is performed the gateway should be in the correct Ethernet setting prior to the time this parameter takes effect When for example the gateway is configured using BootP TFTP the gateway must perform many Ethernet based transactions prior to reading the ini file containing this gateway configuration parameter To work around this problem the gateway always uses the last Ethernet setup mode configured This way if users want to configure the gateway to work in a new network environment in which the current Ethernet setting of the gateway is invalid they should first modify this parameter in the current network so that the new setting holds next time gateway is restarted After reconfiguration has completed connect the gateway to the new network and restart it As a result the remote configuration process that takes place in the new network uses a valid Ethernet configuration NAT Network Address Translation Support 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 are 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 security by hiding its internal architecture Figure 9 1 below illustrates the NAT architecture Figure 9 1 NAT Functioning MediaPack
10. 12 1 Security This section describes the security mechanisms and protocols implemented on the MediaPack The following list specifies the available security protocols and their objectives m PSec and IKE protocols are part of the IETF standards for establishing a secured IP connection between two applications IPSec and IKE are used in conjunction to provide security for control and management protocols but not for media refer to Section 12 1 below m SSL Secure Socket Layer TLS Transport Layer Security The SSL TLS protocols are used to provide privacy and data integrity between two communicating applications over TCP IP They are used to secure the following applications SIP Signaling SIPS Web access HTTPS and Telnet access refer to Section 12 2 on page 288 E Secured RTP SRTP according to RFC 3711 used to encrypt RTP and RTCP transport refer to Section 12 3 on page 293 m RADIUS Remote Authentication Dial In User Service RADIUS server is used to enable multiple user management on a centralized platform refer to Section 12 4 on page 294 Internal Firewall allows filtering unwanted inbound traffic refer to Section 12 5 on page 297 IPSec and IKE IP Security IPSec and Internet Key Exchange IKE protocols are part of the IETF standards for establishing a secured IP connection between two applications also referred to as peers Providing security services at the IP layer IPSec and IKE are tra
11. 2 Configure the RTP RTCP Settings according to Table 5 44 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Version 5 0 157 December 2006 Ta fal AudioCodes MediaPack Table 5 44 Media Settings RTP RTCP Parameters continues on pages 158 to 159 Parameter Dynamic Jitter Buffer Minimum Delay DJBufMinDelay Dynamic Jitter Buffer Optimization Factor DJBufOptFactor RTP Redundancy Depth RTPRedundancyDepth Packing Factor RTPPackingFactor Basic RTP Packet Interval BasicRTPPacketinterval Note This parameter should not be used Use the Coders screen under Protocol Definition instead RTP Directional Control RTPDirectionControl RFC 2833 TX Payload Type RFC2833TxPayloadType RFC 2833 RX Payload Type RFC2833RxPayloadType RFC 2198 Payload Type RFC2198PayloadType Fax Bypass Payload Type FaxBypassPayloadType Enable RFC 3389 CN Payload Type EnableStandardSIDPayloadType Analog Signal Transport Type AnalogSignalTransportType Description Minimum delay for the Dynamic Jitter Buffer The valid range is 0 to 150 milliseconds The default delay is 10 milliseconds Note For more information on the Jitter Buffer refer to Section 8 6 on page 230 Dynamic Jitter Buffer frame error delay optimization factor The valid range is 0 to 13 The default factor
12. 5 Click the Send File button that is next to the field that contains the name of the file you want to load An exclamation mark in the screen section indicates that the file s loading doesn t take effect on the fly e g CPT file 6 Repeat steps 2 to 5 for each file you want to load Saving a configuration file to flash memory may disrupt traffic on the MediaPack To avoid this disable all traffic on the device before saving to flash memory by performing a graceful lock refer to Section 5 10 1 on page 204 A device reset is required to activate a loaded CPT file 7 To save the loaded auxiliary files so they are available after a power fail refer to Section 5 10 2 on page 205 8 To reset the MediaPack refer to Section 5 10 3 on page 206 Version 5 0 167 December 2006 Ta WH wt AudioCodes MediaPack 5 6 5 Security Settings From the Security Settings you can Configure the Web User Accounts refer to Section 5 6 5 1 below Configure the Web amp Telnet Access List refer to Section 5 6 5 2 on page 170 Configure the Firewall Settings refer to Section 5 6 5 3 on page 171 Configure the Certificates refer to Section 5 6 5 4 on page 172 Configure the General Security Settings refer to Section 5 6 5 5 on page 173 Configure the IPSec Table refer to Section 5 6 5 6 on page 175 Configure the IKE Table refer to Section 5 6 5 7 on page 176 5 6 5 1 Configuring the Web User Accounts To prev
13. DES 3DES AES128 AES192 or AES256 Each SNMP v3 user must be associated with one of the predefined groups listed in the following table Table 14 4 SNMP v3 Predefined Groups Group Get Access Set Access Send Traps Security Level ReadGroup1 Yes No Yes noAuthNoPriv 1 ReadWriteGroup1 Yes Yes Yes noAuthNoPriv 1 TrapGroup1 No No Yes noAuthNoPriv 1 ReadGroup2 Yes No Yes authNoPriv 2 ReadWriteGroup2 Yes Yes Yes authNoPriv 2 TrapGroup2 No No Yes authNoPriv 2 ReadGroup3 Yes No Yes authPriv 3 ReadWriteGroup3 Yes Yes Yes authPriv 3 TrapGroup3 No No Yes authPriv 3 Version 5 0 315 December 2006 a WH wt AudioCodes MediaPack 14 8 2 1 Configuring SNMP v3 Users via the ini File Use the SNMPUsers ini table to add modify and delete SNMPv3 users For a description of the SNMPUsers table ini file parameters refer to Section 5 6 6 3 on page 181 Note The SNMPUsers ini table is a hidden parameter Therefore when you perform a Get ini File operation using the Web interface the table will not be included in the generated file You can enter keys in the form of a text password or in the form of a localized key in hex format If using a text password then it should be at least eight characters in length Below is an example of a localized key format 20 00 08 7d Uds4a do Se 02 7S 0d 22 906 e72 69 08 The following example configuration creates three SNMPv3 USM users SNMPUsers FOR
14. DisableAutoDTMFMute 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 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 Determines the index of the first Call Waiting Tone in the CPT file This feature enables the called party to distinguish between four different call origins e g external vs internal calls The gateway plays the tone received in the play tone CallWaitingTone parameter of an INFO message the value of this parameter 1 The valid range is 1 to 100 The default value is 1 not used Note 1 It is assumed that all Call Waiting Tones are defined in sequence in the CPT file Note 2 This feature is relevant only to Broadsoft s application servers the tone is played using INFO message Determines the FXO line characteristics AC and DC according to country of origin Argentina 0 Australia 1 Austria 2 Bahrain 3 Belgium 4 Brazil 5 Bulgaria 6 Canada 7 Chile 8 China 9 Colombia 10 Croatia 11 Cyprus 12 Czech_Republic 13 Denmark 14 Ecuador 15 Egypt 16 El_ Salvador 17 Finland 18 France 19 Germany 20 Greece 21 Guam 22 Hong_Kong 23 Hungary 24 Iceland 25 India 26 Indonesia 27 Ireland 2
15. E PM Analog acPMAnalogConfigurationResetT otalCounters PM Control acPMControlConfigurationResetTotalCounters PM Media acPMMediaConfigurationResetTotalCounters PM PSTN acPMPSTNConfigurationResetT otalCounters PM System acPMSystemConfigurationResetTotalCounters Supported MIBs The MediaPack contains an embedded SNMP Agent supporting the following MIBs E Standard MIB MIB 2 The various SNMP values in the standard MIB are defined in RFC 1213 The standard MIB includes various objects to measure and monitor IP activity TCP activity UDP activity IP routes TCP connections interfaces and general system indicators RTP MIB The RTP MIB is supported in conformance with the IETF RFC 2959 It contains objects relevant to the RTP streams generated and terminated by the device and to RTCP information related to these streams E NOTIFICATION LOG MIB This standard MIB RFC 3014 iso org dod internet mgmt mib 2 is supported as part of our implementation of carrier grade alarms ALARM MIB This is an IETF MIB RFC 3877 also supported as part of our implementation of carrier grade alarms This MIB is a new standard and is therefore under the audioCodes acExperimental branch SNMP TARGET MIB According to RFC 2273 It allows for the configuration of trap destinations and trusted managers SNMP MIB This MIB RFC 3418 allows support of the coldStart and authenticationFailure traps SNMP Framework MIB RFC 3411 SNMP
16. SIP User s Manual 50 Document LTRT 65408 SIP User s Manual 5 Web Management 5 2 2 Limiting the Embedded Web Server to Read Only Mode Users can limit access to the Embedded Web Server to read only mode by changing the ini file parameter DisableWebConfig to 1 In this mode all Web screens regardless of the access level used are read only and cannot be modified In addition the following screens cannot be accessed Quick Setup Web User Accounts Reset Save Configuration and all of the file loading screens Notes e Read only policy can also be applied to selected users by setting the access level of the secondary account to User Monitor DisableWebConfig 0 and distributing the primary and secondary accounts to users according to the organization s security policy e When DisableWebConfig is set to 1 read only privileges are applied to all accounts regardless of their access level 5 2 3 Disabling the Embedded Web Server Access to the Embedded Web Server can be disabled by using the ini file parameter DisableWebTask 1 The default is access enabled 5 3 Accessing the Embedded Web Server gt To access the Embedded Web Server take these 4 steps 1 Open a standard Web browsing application such as Microsoft Internet Explorer or Netscape Navigator 2 In the Uniform Resource Locator URL field specify the IP address of the MediaPack e g http 10 1
17. Supporting V 34 Faxes V 34 faxes don t comply with the T 38 relay standard We therefore provide the optional modes described under Sections 8 3 4 1 and 8 3 4 2 Note that the CNG detector is disabled CNGDetectorMode 0 in all the following examples Using Bypass Mechanism for V 34 Fax Transmission In this proprietary scenario the media gateway uses a high bit rate coder to transmit V 34 faxes enabling the full utilization of its speed Refer to the following configurations FaxTransportMode 2 Bypass V34ModemTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemTransportType 2 V22ModemTransportType 2 In this configuration both T 30 and V 34 faxes work in Bypass mode Or FaxTransportMode 1 Relay V34ModemTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemTransportType 2 V22ModemTransportType 2 8 3 4 2 In this configuration T 30 fax uses T 38 Relay mode while V 34 fax uses Bypass mode Using Relay mode for both T 30 and V 34 faxes In this scenario V 34 fax machines are compelled to use their backward compatibility with T 30 faxes as a T 30 machine the V 34 fax can use T 38 Relay mode Refer to the following configuration FaxTransportMode 1 Relay V34ModemTransportType 0 Transparent V32ModemTransportType 0 V23ModemTransportType 0 V22ModemTransportType 0 Both T 30 and V 34 faxes use T 38 Relay mode This co
18. answers the call and sends 200 OK message to gateway 10 8 201 158 F3 10 8 201 161 gt 10 8 201 158 200 OK STE 2 0 ZOO G Via SIP 2 0 UDP 10 8 201 158 branch z9hG4bKacolwbzYF From lt sip 2000 10 8 201 158 gt tag 1c3535 ros lt eo G O00 CO e 0 leleG let cealc ike 2 Olas Call ips 212335377537 Nrpe 2000 _s000010 2 201 15 CSeq 20214 INVITE Contact lt sip 6000 10 8 201 161 user phone gt Server Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 208 v 0 s Phone Call t 0 0 o AudiocodesGW 30762 37542 IN IP4 10 8 201 161 c IN TEM 10 8 201 161 m audio 4040 RTP AVP 8 96 a rtpmap 8 pcma 8000 a ptime 20 a rtpmap 96 telephone event 8000 a fmtp 96 0 15 F4 10 8 201 158 gt 10 8 201 161 ACK ACK sip 6000 10 8 201 161 user phone user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 158 branch z 9hG4bKachoWSQxD From lt sip 2000 10 8 201 158 gt tag 1c3535 To lt eo HG OO0ChOmMs ra Ole Git reag ike 2 oes Call ID 2123353775377NrpL 2000 6000 10 8 201 158 User Agent Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 CSeq 20214 ACK Supported 100rel em Content Length 0 Note Phone 6000 goes onhook gateway 10 8 201 161 sends BYE to gateway 10 8 201 158 Voice path is established Version 5 0 239 December 2006 Ta
19. 1 gt annexb yes EnableSilenceCompression 2 and IsCiscoSCEMode 0 gt annexb yes EnableSilenceCompression 2 and IsCiscoSCEMode 1 gt annexb no Off 0 Echo Canceler disabled On 1 Echo Canceler enabled default DTMF Mute remote Transparent DTMF 2 Digits remain in voice stream RFC 2833 Relay DTMF 3 Erase digits from voice stream relay to remote according to RFC 2833 default Note This parameter is automatically updated if one of the following parameters is configured TXxDTMFOption or RxDTMF Option 0 Erase digits from voice stream do not relay to N A DTMF gain control value in dB to the analog side The valid range is 31 to 0 dB The default value is 11 dB N A N A N A N A Determines the Answer Detector sensitivity The range is 0 most sensitive to 2 least sensitive The default is 0 Version 5 0 153 December 2006 Ta A wl AudioCodes MediaPack 5 6 2 2 Configuring the Fax Modem CID Settings gt To configure the Fax Modem CID Settings parameters take these 4 steps 1 Open the Fax Modem CID Settings screen Advanced Configuration menu gt Media Settings gt Fax Modem CID Settings option the Fax Modem CID Settings screen is displayed Figure 5 41 Fax Modem CID Settings Screen Fax Modem CID Settings Fax Transport Mode M38 Rey H Caller ID Transport Type Mte Caller ID Type Be
20. 1 SRV 2 NAPTR If set to A Record no NAPTR or SRV queries are performed If set to SRV and the Proxy Registrar IP address parameter or the domain name in the Contact Record Route headers contains a domain name without port definition an SRV query is performed The gateway uses the first host name received from the SRV query The gateway then performs DNS A record query for the host name to locate an IP address If set to NAPTR 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 or the domain name in the Contact Record Route headers contains a domain name with port definition the gateway performs a regular DNS A record query Note To enable NAPTR SRV queries only for Proxy servers use the parameter ProxyDNSQueryType This parameter is obsolete Please use the parameter DNSQueryType SIP User s Manual 68 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 4 Proxy amp Registration Parameters continues on pages 66 to 71 Parameter Description Proxy DNS Query Type Enables the use of DNS Naming Authority Pointer NAPTR and Service Record ProxyDNSQueryType SRV queries to discover Proxy servers Valid options include 0 A Record default 1 SRV 2 NAPTR If set t
21. 14 8 1 3 Configuration of Community Strings via SNMP To configure read only and read write community strings the EM must use the SNMP COMMUNITY MIB To configure the trap community string the EM must also use the snmpVacmMIB and the snmpTargetMIB gt a To add a read only community string v2user take this step Add a new row to the srCommunityTable with CommunityName v2user and GroupName ReadGroup To delete the read only community string v2user take these 2 steps If v2user is being used as the trap community string follow the procedure for changing the trap community string see below Delete the ssCommunityTable row with CommunityName v2user To add a read write community string v2admin take this step Add a new row to the srCommunityTable with CommunityName of v2admin and GroupName ReadWriteGroup To delete the read write community string v2admin take these 2 steps If v2admin is being used as the trap community string follow the procedure for changing the trap community string See below Delete the srCommunityTable row with a CommunityName of v2admin and GroupName of ReadWriteGroup To change the only read write community string from v2admin to v2mgr take these 4 steps Follow the procedure above to add a read write community string to a row for v2mgr Set up the EM so that subsequent set requests use the new community string v2mgr If v2admin is being used as the trap community strin
22. 4 auto negotiate default For detailed information on Ethernet interface configuration refer to Section 9 1 on page 247 This parameter is used to Note This switch takes effect only from the next gateway reset Set the number of BootP requests the gateway Set the number of DHCP packets the gateway sends during start up The gateway stops sends sending BootP requests when either BootP reply After all packets were sent if there s still no reply is received or number of retries is reached the gateway loads from flash 1 1 BootP retry 1 second 1 4 DHCP packets 2 2 BootP retries 3 seconds 2 5 DHCP packets 3 3 BootP retries 6 seconds 3 6 DHCP packets default 4 10 BootP retries 30 seconds 4 7 DHCP packets 5 20 BootP retries 60 seconds 5 8 DHCP packets 6 40 BootP retries 120 seconds 6 9 DHCP packets 7 100 BootP retries 300 seconds 7 10 DHCP packets 15 BootP retries indefinitely 15 18 DHCP packets Use bs 1 to enable the Selective BootP mechanism Use bs 0 to disable the Selective BootP mechanism The Selective BootP mechanism available from Boot version 1 92 enables the gateway 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 gateway s BootP requests Us
23. AccessList_Source_IP AccessList_Net_Mask AccessList_Start_Port AccessList_End_Port AccessList_Protocol AccessList_Packet_Size AccessList_Byte_Rate AccessList_Byte_Burst AccessList_Allow_Type MCSSsihileic MO mene Cuigionise com 255 255 255 255 OW 80 eed WO O MW Aer F Agcesclust el 5 eih92 100m 0 eez oo 0 On 0 uO meOO oS oi any O OO 00 Fas 00 00pm blocks MecCesebiec 20 0 S 0 255 255 25959 0 OOO GOOW aim OW OF Wy lalkoele p Accessbist 22 KOMTO 255 29 Or Ome O0O me S000 any O Op O block VACCHESSLIST Explanation of the example access list m Rule 10 traffic from the host mgmt customer com destined to TCP ports 0 to 80 is always allowed m Rule 15 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 Rule 20 traffic from the subnet 10 31 4 xxx destined to ports 4000 to 9000 is always blocked regardless of protocol E Rule 22 traffic from the subnet 10 4 xxx yyy destined to ports 4000 to 9000 is always blocked regardless of protocol E All other traffic is allowed More complex rules may be defined relying on the single match process described above Figure 12 16 shows an advanced example of an access list definition via ini file Figure 12 16
24. For example ConferencelD MyConference 5 5 2 3 Metering Tones FXS gateways 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 SIP User s Manual 86 Document LTRT 65408 SIP User s Manual 5 Web Management To configure the Metering Tones take these 6 steps Open the Metering Tones screen Protocol Management menu gt Advanced Parameters submenu gt Metering Tones option the Metering Tones screen is displayed Figure 5 11 Metering Tones Parameters Screen Metering Tones Generate Metering Tones Disable Metering Tone Type 12 KHz Parameter Charge Codes Table o gt From the Generate Metering Tones drop down list select the method used to configure the metering tones that are generated to the Tel side refer to Table 5 10 If you selected Internal Table you must configure the Charge Codes Table To configure the Charge Codes Table refer to Section 5 5 2 3 1 below Continue with Step 4 From the Metering Tone Type drop down list select the type of the metering tone according
25. LTRT 65408 SIP User s Manual 5 Web Management Figure 5 64 Load a cmp File Screen Load a CMP file frorn your computer to the device Browse 4 Click the Browse button navigate to the cmp file and click the button Send File the cmp file is loaded to the MediaPack and you re notified as to a successful loading refer to Figure 5 65 Figure 5 65 cmp File Successfully Loaded into the MediaPack Notification INI file CPT file PRT file File MP118_SIP_F5 00A 001 003 cmp was successfully Finish loaded into the device 5 Note that the four action buttons Cancel Reset Back and Next are now activated following cmp file loading 6 You can now choose to either e Click Reset the MediaPack resets utilizing the new cmp you loaded and utilizing the current configuration files e Click Cancel the MediaPack resets utilizing the cmp ini and all other configuration files that were previously stored in flash memory Note that these are NOT the files you loaded in the previous Wizard steps e Click Back the Load a cmp File screen is reverted to refer to Figure 5 64 e Click Next the Load an ini File screen opens refer to Figure 5 66 Loading a new ini file or any other auxiliary file listed in the Wizard is optional Note that as you progress the file type list on the left indicates which file type loading is in process by illuminating green until FINISH Version 5 0 199 Dec
26. SNMPTrapManagerHostName configure the trap manger host For example name use the parameter SNMPTrapManagerHostName myMananger corp MyCompany com Note The same information configurable in the ini file can also be configured via the acBoardMIB SIP User s Manual 320 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 8 5 3 Configuring Trap Managers via SNMP The standard snmpTargetMIB interface is available for configuring trap managers Note The acBoard MIB is planned to become obsolete The only relevant section in this MIB is the trap sub tree acTrap To add an SNMPvz2 trap destination take the following step Add a row to the snmpTargetAddrTable with these values Name trapN TagList AC_TRAP Params v2cparams where N is an unused number between 0 and 4 All changes to the trap destination configuration take effect immediately To add an SNMPv3 trap destination take these 2 steps Add a row to the snmpTargetAddrTable with these values o Name trapN where N is an unused number between 0 and 4 e TagList AC_TRAP Params usm lt user gt where lt user gt is the name of the SNMPv3 with which this user is associated If a row does not already exist for this combination of user and SecurityLevel add a row to the snmpTargetParamsTable with these values e Name usm lt user gt e MPModel 3 SNMPv3 e SecurityModel 3 usm e SecurityName lt user gt e Securi
27. STUN Server Secondary IP STUNServerSecondaryIP NFS Settings NFS Table IP address in dotted format notation of the NTP server The default IP address is 0 0 0 0 the internal NTP client is disabled Defines the UTC Universal Time Coordinate offset in seconds from the NTP server The default offset is 0 The offset range is 43200 to 43200 seconds Defines the time interval in seconds the NTP client requests for a time update The default interval is 86400 seconds 24 hours The range is 0 to 214783647 seconds Note It isn t recommended to be set beyond one month 2592000 seconds Enables or disables the embedded Telnet server Telnet is disabled by default for security reasons Disable 0 default Enable Unsecured 1 Enable Secured SSL 2 N A Defines the port number for the embedded Telnet server The valid range valid port numbers The default port is 23 Sets the timeout for disconnection of an idle Telnet session in minutes When set to zero idle sessions are not disconnected The valid range is any value The default value is 0 Disable 0 STUN protocol is disabled default Enable 1 STUN protocol is enabled When enabled the gateway functions as a STUN client and communicates with a STUN server located in the open internet STUN is used to discover whether the gateway is located behind a NAT and the type of that NAT In addition it is used to determine the IP addresses and p
28. The MediaPack accommodates an internal access list facility allowing the security administrator to define network traffic filtering rules The access list provides the following features E Block traffic from known malicious sources Only allow traffic from known friendly sources and block all others m Mix allowed and blocked network sources E Limit traffic to a predefined rate blocking the excess E Limit traffic to specific protocols and specific port ranges on the device The access list consists of a table with up to 50 ordered lines For each packet received on the network interface the table is scanned from the top until a matching rule is found or the table end is reached This rule can either block the packet or allow it however it is important to note that subsequent rules aren t scanned If the table end is reached without a match the packet is accepted Each rule is composed of the following fields described in Table 5 49 on page 172 IP address or DNS name of source network IP network mask Destination UDP TCP ports on this device Protocol type Maximum packet size byte rate per second and allowed data burst Action upon match allow or block Version 5 0 297 December 2006 Ta WH wt AudioCodes MediaPack Figure 12 15 shows an example of an access list definition via ini file Figure 12 15 Example of an Access List Definition via ini File ACCESSLIST FORMAT AccessList_Index
29. Unpack the MP 11x refer to Section 3 1 1 Check the package contents refer to Section 3 1 2 Mount the MP 11x refer to Section 3 1 4 on page 30 Cable the MP 11x refer to Section 3 1 5 on page 38 After connecting the MP 11x to the power source the Ready and Power LEDs on the front panel turn to green after a self testing period of about two minutes Any malfunction in the startup procedure changes the Fail LED to red and the Ready LED is turned off refer to Table 2 1 on page 26 for details on the MP 11x LEDs When you have completed the above relevant sections you are then ready to start configuring the gateway Chapter 5 on page 49 ee 3 1 1 Unpacking To unpack the MP 11x take these 6 steps Open the carton and remove the packing materials Remove the MP 11x gateway from the carton Check that there is no equipment damage Check retain and process any documents Notify AudioCodes or your local supplier of any damage or discrepancies Retain any diskettes or CDs nee Sy 3 1 2 Package Contents Ensure that in addition to the MP 11x the package contains E AC power cable Small plastic bag containing four anti slide bumpers for desktop installation A CD with software and documentation may be included m The MediaPack Fast Track Installation Guide Version 5 0 29 December 2006 Ta A wi AudioCodes MediaPack 3 1 3 19 inch Rack Installation Package Optional An additional optio
30. WH wt AudioCodes MediaPack 13 3 2 Operation The Syslog client embedded in the MediaPack sends error reports events generated by the MediaPack unit application to a Syslog server using IP UDP protocol gt 1 2 To activate the Syslog client on the MediaPack take these 5 steps Set the parameter EnableSyslog to 1 refer to Table 5 51 on page 177 Use the parameter SyslogServerIP to define the IP address of the Syslog server you use refer to Table 5 51 on page 177 Use the parameter SyslogServerPort to define the UDP port number of the Syslog server refer to Table 5 51 on page 177 To determine the Syslog logging level use the parameter GWDebugLevel refer to Table 5 8 on page 78 To enable the gateway to send log messages that report certain types of Web actions according to a pre defined filter use the parameter ActivityListToLog described in Table 5 51 on page 177 SIP User s Manual 304 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 14 1 14 1 1 SNMP Based Management Simple Network Management Protocol SNMP is a standard based network control protocol used to manage elements in a network The SNMP Manager usually implemented by a Network Manager NM or an Element Manager EM connects to an SNMP Agent embedded on a remote Network Element NE to perform network element Operation Administration and Maintenance OAM Both the SNMP Man
31. period in 10 msec units for the first cadence on off cycle For be continuous tones this parameter defines the detection period For burst tones it defines the tone s duration 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 ends and the tone detection is reported For continuous tones this parameter is ignored 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 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 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 SIP User s Manual 326 Document LTRT 65408 SIP User s Manual 15 Configuration Files 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 Forth 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 Forth Signal Off Time 10 msec Signal Off period in 10 msec units
32. 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 gateway s BootP requests Note When working with DHCP DHCPEnable 1 the selective BootP feature must be disabled 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 This parameter only takes effect from the next reset of the device 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 board type current IP address software version etc For a full list of the vendor specific Information fields refer to Section 7 3 2 on page 214 The BootP TFTP configuration utility displays this information in the Client Info column refer to Figure C 1 Note This option is not available on DHCP servers 5 6 6 5 Automatic Updates Parameters For detailed information on the automatic update mechanism refer to Section 10 3 on page 263 Table 5 56 Automatic Updates Parameters continues on pages 184 to 185 ini File Parameter Name D
33. sending INVITE message to Gateway 10 8 201 161 Figure 8 6 SIP Call Flow 10 8 201 158 10 8 201 161 INVITE Ringing F1 10 8 201 158 gt 10 8 201 161 INVITE INVITE sip 6000 10 8 201 161 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 158 branch z9hG4bKacolwbzYF From lt sip 2000 10 8 201 158 gt tag 1c3535 Tos Swor HOMOCE Ost AOI Weis Call iD 2123353775371 NepkL 2000 conve los col 156 CSeq 20214 INVITE Contact lt sip 2000 10 8 201 158 user phone gt User Agent Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 208 v 0 s Phone Call t 0 0 o AudiocodesGW 87943 43401 IN IP4 10 8 201 158 c IN IP4 10 8 201 158 m audio 6000 RTP AVP 8 96 a rtpmap 8 pcma 8000 a rtpmap 96 telephone event 8000 a fmtp 96 0 15 a ptime 20 SIP User s Manual 238 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities F2 10 8 201 161 gt 10 8 201 158 180 RINGING SIP 2 0 180 Ringing Via SIP 2 0 UDP 10 8 201 158 branch z9hG4bKacolwbzYF WHOS lt Sajoe2OOOEIO 6 201 eS pcscHle 3535 io lt sap 7600 0CL0Mse 20s LG eag le2 SS Call ID 2123353775377NrpL 2000 6000 10 8 201 158 Server Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 CSeq 20214 INVITE Supported 100rel em Content Length 0 Note Phone 2000
34. 1 PassThrough 2 N A Note This parameter cannot be changed on the fly and requires a gateway reset Sets the native VLAN identifier PVID Port VLAN ID The valid range is 1 to 4094 The default value is 1 Sets the OAM Operation Administration and Management VLAN identifier The valid range is 1 to 4094 The default value is 1 Sets the control VLAN identifier The valid range is 1 to 4094 The default value is 2 Sets the media VLAN identifier The valid range is 1 to 4094 The default value is 3 Sets the priority for Network service class content The valid range is 0 to 7 The default value is 7 Sets the priority for the Premium service class content and media traffic The valid range is 0 to 7 The default value is 6 Sets the priority for the Premium service class content and control traffic The valid range is 0 to 7 The default value is 6 Sets the priority for the Gold service class content The valid range is 0 to 7 The default value is 4 Sets the priority for the Bronze service class content The valid range is 0 to 7 The default value is 2 Determines the traffic type for DNS services 1 OAM VLAN default 0 Control VLAN Determines the traffic type for NTP services 1 OAM VLAN default 0 Control VLAN Specify whether to send non tagged packets on the native VLAN 0 Sends priority tag packets default 1 Sends regular packets with no VLAN tag SIP User
35. Dial Go on hook p Start Recording Stop Recording Play Tone type Dial Tone Analyze recorded tone IV Play through Status The gateway detected Dial Tone Tones analyzed ToneType LoFreg Hi Freg 1st On 1st Off 2nd On 2nd Off Detected Dial Tone auto 350 440 0 0 N Busy Tone aut 480 620 50 Ringing Tone 450 500 180 Reorder Tone 480 620 25 fa AudioCodes Cancel 2 Check the play through check box to hear the tones through your PC speakers 3 Click the Go offhook button enter a number to dial in the Dial field and click the Dial button When you re ready to record click the Start Recording button when the desired tone is complete click Stop Recording The recorded tone is saved as cpt_manual_tone pcm Note Due to some PC audio hardware limitations you may hear clicks in play through mode It is safe to ignore these clicks 4 Select the tone type from the drop down list and click Analyze recorded tone the analyzed tone is added to the Tones analyzed list at the bottom of the screen It is possible to record and analyze several different tones for the same tone type e g different types of busy signal 5 Repeat the process for more tones as necessary 6 When you re finished adding tones to the list click Next to generate a Call Progress Tones ini file and terminate the wizard Version 5 0 371 December 2006 Ta WH wt Audi
36. Enable SMDI SMDI SMDI Timeout SMDITimeOut Determines the digit pattern used by the PBX to indicate call forward on busy The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on no answer The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on do not disturb The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward with no reason 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 gateway to disconnect the call The valid range is a 25 character string Determines a digit code used by the gateway 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 a digit code used by the gateway 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 Enables the Simplified Message Desk Interface SMDI
37. For example 404 for 3 503 for 34 and 502 for 27 Defines the amount of time in seconds the gateway s operation is delayed after a reset cycle The valid range is 0 to 45 The default value is 7 seconds Note This feature helps to overcome connection problems caused by some LAN routers or IP configuration parameters change by a DHCP Server Defines the maximum number of calls that the gateway can have active at the same time If the maximum number of calls is reached new calls are not established The default value is max available channels no restriction on the maximum number of calls The valid range is 1 to max number of channels Defines the maximum call duration in seconds If this time expires both sides of the call are released IP and Tel The valid range is 0 to 120 The default is O no limitation Disable 0 Disable LAN Watch Dog default Enable 1 Enable LAN Watch Dog When LAN Watch Dog is enabled the gateway s overall communication integrity is checked periodically If no communication for about 3 minutes is detected the gateway performs a self test If the self test succeeds the problem is logical link down i e Ethernet cable disconnected on the switch side and the Busy out mechanism is activated if enabled EnableBusyOut 1 LifeLine is activated if enabled If the self test fails the gateway restarts to overcome internal fatal communication error Note Enable LAN Watchdog is relevan
38. In addition this parameter defines the time interval between Keep Alive messages when EnableProxyKeepAlive 2 REGISTER Typically a value of 3600 should be assigned for one hour registration The gateway resumes registration according to the parameter RegistrationTimeDivider The default value is 180 The valid range is 10 to 2000000 Version 5 0 69 December 2006 Ta WH wt AudioCodes MediaPack Table 5 4 Proxy amp Registration Parameters continues on pages 66 to 71 Parameter Re registration Timing RegistrationTimeDivide r Registration Retry Time RegistrationRetryTime Subscription Mode SubscriptionMode Enable Proxy Keep Alive EnableProxyKeepAlive Proxy Keep Alive Time ProxyKeepAliveTime Use Gateway Name for OPTIONS UseGatewayNameForO ptions Enable Fallback to Routing Table IsFallbackUsed Prefer Routing Table PreferRouteTable Description Defines the re registration timing in percentage The timing is a percentage of the re register timing set by the Registration server The valid range is 50 to 100 The default value is 50 For example If RegistrationTimeDivider 70 and Registration Expires time 3600 the gateway resends its registration request after 3600 x 70 2520 sec Defines the time period in seconds after which a Registration request is resent if registration fails with 4xx or there is no response from the Proxy Registrar The default
39. LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 12 Configuration Examples 8 12 1 Establishing a Call between Two Gateways After you ve installed and set up the MediaPack you can ensure that it functions as expected by establishing a call between it and another gateway This section exemplifies how to configure two 8 port MediaPack FXS SIP gateways to establish a call After configuration you can make calls between telephones connected to a single MediaPack gateway or between the two MediaPack gateways In the following example the IP address of the first gateway is 10 2 37 10 and its endpoint numbers are 101 to 108 The IP address of the second gateway is 10 2 37 20 and its endpoint numbers are 201 to 208 In this example a SIP Proxy is not used Internal call routing is performed using the internal Tel to IP Routing table gt To configure the two gateways take these 4 steps 1 For the first MediaPack gateway 10 2 37 10 in the Endpoint Phone Numbers screen assign the phone numbers 101 to 108 for the gateway s endpoints Endpoint Phone Number Table Channel s Phone Number__Hunt Group ID_ Profile ID __ 1 FXS 1 8 101 2 For the second MediaPack gateway 10 2 37 20 in the Endpoint Phone Numbers screen assign the phone numbers 201 to 208 for the gateway s endpoints Endpoint Phone Number Table Channel s Phone Number Hunt Group ID Profile ID 1 8
40. Note Table 10 1 and Table 10 2 are provided as examples for the purpose of illustration only and are not actually implemented in the MediaPack 10 5 1 Table Indices Each line in a table must be unique Therefore each table defines one or more Index fields The combination of the Index fields determines the line tag Each line tag appears only once In the example provided in Table 10 1 there is only one Index field This is the simplest way to mark lines In the example provided in Table 10 2 there are three Index fields This more complicated method is a result of the application it represents Version 5 0 267 December 2006 7a Ta r wt AudioCodes MediaPack 10 5 2 10 5 3 10 5 4 Table Permissions Each column has a permission attribute that is applied to all instances in the column This permission determines if and when a field can be modified Several permissions can be applied to each column The following permissions are available m Read The value of the field can be read E Write The value of the field can be modified m Create A value for the field must be provided at creation time the default values set to all fields determine the initial values Maintenance Write The value of the field can only be modified when the entity represented by the line is in maintenance state each table includes rules that determine when it is in maintenance state In the example in Table 10 1 it is assume
41. Port Number 23 68 80 161 443 500 6000 6010 and up 6001 6011 and up 6002 6012 and up 5060 5061 random gt 32767 random gt 32767 random gt 32767 random gt 32767 random gt 32767 Table 12 5 Default TCP UDP Network Port Numbers Peer Port 67 5060 5061 514 162 Application Debugging interface Telnet DHCP Web server HTTP SNMP GET SET Web server HTTPS IPSec IKE RTP traffic RTCP traffic T 38 traffic SIP SIP over TLS SIPS Syslog Syslog ICMP ARP listener SNMP Traps DNS client Notes Always ignored Disabled by default TelnetServerEnable Configurable TelnetServerPort access controlled by WebAccessList Active only if DHCPEnable 1 Configurable HTTPPort can be disabled DisableWebTask or HTTPSOnly Access controlled by WebAccessList Configurable SNMPPort can be disabled DisableSNMP Access controlled by SNMPTrustedMGR Configurable HTTPSPort can be disabled DisableWebTask Access controlled by WebAccess_List Can be disabled EnablelPSec Base port number configurable BaseUDPPort fixed increments of 10 The number of ports used depends on the channel capacity of the device Always adjacent to the RTP port number Always adjacent to the RTCP port number Configurable LocalSIPPort UDP TCPLocalSIPPort TCP Configurable TLSLocalSIPPort Disabled by default EnableSyslog Disabled by default EnableSy
42. RouteModeTel2IP manipulation rules are applied default Route calls after manipulation 1 Tel gt IP calls are routed after the number manipulation rules are applied Note Not applicable if Proxy routing is used Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers Any telephone number whose destination number matches the prefix defined in the Destination Phone Prefix field and its source number matches the prefix defined in the adjacent Source Phone Prefix field is sent to the IP address entered in the IP Address field Note that Tel to IP routing can be performed according to a combination of source and destination phone prefixes or using each independently Note 1 An additional entry of the same prefixes can be assigned to enable alternative routing Note 2 For available notations that represent multiple numbers refer to Section 5 5 3 1 on page 95 Version 5 0 101 December 2006 Ta WH wt AudioCodes MediaPack Table 5 17 Tel to IP Routing Table continues on pages 101 to 102 Parameter Description Destination IP Address In each of the IP Address fields
43. The lines were tested under the following conditions ring voltage greater than 30 Vrms offhook loop current greater than 20 mA all lines ring simultaneously MP 11x includes lightning and high voltage protection for outdoor operation Caller ID generation 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 Programmable Line Characteristics Battery feed line current hook thresholds AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains Note For a specific coefficient file contact AudioCodes Programmable ringing signal Up to three cadences and frequency 15 to 200 Hz Over temperature protection for abnormal situations as shorted lines Loop backs for testing and maintenance MP 11x FXO Functionality FXO Capabilities Note doesn t apply to the MP 112 Additional Features Polarity Reversal Wink Metering Tones Short or Long Haul Includes lightning and high voltage protection for outdoor operation Programmable Line Characteristics AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains ring detection threshold DC characteristics Note For country specific coefficients use the parameter CountryCoefficients Caller ID detection Bellcore GR 30 CORE Type 1 using Bell 202 FSK modulation ETSI Type 1 NTT Denmark India Brazil British and DTMF ETSI
44. Therefore the used NCP will be IPCP IP Configuration Protocol In this phase if the ini file parameter PPPoEStaticlPAddress is defined the gateway requests the remote host to assign this address for its use Version 5 0 251 December 2006 Ta WH wt AudioCodes MediaPack 9 5 9 6 When working in a PPPoE environment the gateway negotiates for its IP address as described above However if the user desires to disable the PPPoE client the gateway can be configured to use default values for IP address subnet mask and default gateway This can be done using ini file parameters PPPoERecoverlPAddress PPPoERecoverSubnetMask and PPPoERecoverDfgwAddress These parameters indicate to the gateway that if the PPPoE is disabled and no BOOTP server is activated as required in the gateway to use a PPPoE environment then the gateway should use these defaults for its IP configuration For a detailed description of the ini file parameters for PPPoE refer to Section 5 6 1 6 on page 149 When working with a PPPoE server Access Concentrator that does not reply to LCP Echo messages which by default the gateway periodically sends you may want to disable the LCP Echo messages by using the ini file parameter PPPoELcpEchoEnable For a description of this parameter refer to Section 5 6 1 6 on page 149 Robust Reception of RTP Streams This mechanism filters out unwanted RTP streams that are sent to the same port number on
45. amp Registration Parameters continues on pages 66 to 71 Parameter Use Routing Table for Host Names and Profiles AlwaysUseRouteTable Always Use Proxy AlwaysSendToProxy Send All INVITE to Proxy SendInviteToProxy Enable Proxy Hot Swap IsProxyHotSwap Number of RTX Before Hot Swap ProxyHotSwapRitx User Name UserName Note The Authentication table can be used instead Password Password Cnonce Cnonce Authentication Mode AuthenticationMode Description Use the internal Tel to IP routing table to obtain the URI Host name and optionally an IP profile per call even if Proxy server is used No 0 Don t use default Yes 1 Use Note This domain name is used instead of Proxy name or Proxy IP address in the INVITE SIP URI No 0 Use standard SIP routing rules default Yes 1 All SIP messages and Responses are sent to Proxy server Note Applicable only if Proxy server is used No 0 INVITE messages 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 30x response default Yes 1 All INVITE messages including those generated as a result of Transfer or Redirect are sent to Proxy Note Applicable only if Proxy server is used and AlwaysSendtoProxy 0 Enable Proxy Hot Swap redundancy mode No 0 Disabled default Yes 1 Enabled If Hot Swap is
46. and profiles to the VoIP gateway ports gt To configure the Endpoint Phone Number table take these 4 steps 1 Open the Endpoint Phone Number Table screen Protocol Management menu gt Endpoint Phone Numbers the Endpoint Phone Number Table screen is displayed Figure 5 25 Endpoint Phone Number Table Screen Endpoint Phone Number Table Channel s Phone Number Hunt Group ID Profile ID 1 4 1 1 2 Configure the endpoint phone numbers according to Table 5 25 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 so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 25 Endpoint Phone Number Table continues on pages 115 to 116 Parameter Description Channel s The numbers 1 8 in the Channel s fields represent the ports on the back of the VoIP gateway To enable a VoIP gateway channel you must enter the port number on this screen n m represents a range of ports For example enter 1 4 to specify the ports from 1 to 4 Phone Number In each of the Phone Number fields enter the telephone number that is assigned to that channel For a range of channels enter the first number in an ordered seq
47. environments The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 10 4 10 4 Subnet Mask to use when booting from the flash to non PPPoE Point to Point Protocol over Ethernet environments The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 255 255 0 0 Default GW address to use when booting from the flash to non PPPoE Point to Point Protocol over Ethernet environments The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 10 4 10 1 Enables or disables the Point to Point Protocol over Ethernet PPPoE disconnection auto detection feature Valid options include 0 Disable 1 Enable default By default the PPPoE Client i e embedded in the gateway sends LCP Echo packets to the server to check that the PPPoE connection is open Some Access Concentrators PPPoE servers don t reply to these LCP Echo requests resulting in a disconnection By disabling the LCP disconnection auto detection feature the PPPoE Client does not send LCP Echo packets to the server and does not detect PPPoE disconnections 0 10 Base T half duplex 1 10 Base T full duplex 2 100 Base TX half duplex 3 100 Base TX full duplex 4 Auto Negotiate default For detailed information on Ethernet interface configuration refer to Section 9 1 on page 247 Version 5 0 149 December 2006 Ta fa AudioCodes Medi
48. for the fourth cadence ON OFF cycle Can be omitted if there isn t a fourth cadence e Carrier Freq Hz the frequency of the carrier signal for AM tones e Modulation Freq Hz the frequency of the modulated signal for AM tones valid range from 1 Hz to 128 Hz e Signal Level dBm the level of the tone for AM tones e AM Factor steps of 0 02 the 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 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 should differ by at least 40 Hz from one tone to other defined tones For example to configure the dial tone to 440 Hz only define the following text Figure 15 2 Defining a Dial Tone Example Dial tone CALL PROGRESS TONE 1 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 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 Figure 15 3 Example of Ringing Burst Three ringing bursts f
49. on the gateway Disable 0 Normal serial default Enable 1 Enable RS 232 SMDI interface Note 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 Determines the time in msec that the gateway waits for an SMDI Call Status message before or after a Setup message is received This parameter is used to synchronize the SMDI and analog interfaces If the timeout expires and only an SMDI message was received the SMDI message is dropped If the timeout expires and only a Setup message was received the call is established The valid range is 0 to 10000 10 seconds The default value is 2000 Version 5 0 131 December 2006 7a Ta r wt AudioCodes MediaPack 5 5 13 Protocol Management ini File Parameters Table 5 35 describes the SIP Protocol Management parameters that can only be configured via the ini file Table 5 35 Protocol Management ini File Parameters continues on pages 132 to 137 ini File Parameter Name EnablePtime IsUseToHeaderAsCalled Number SIPSRequireClientCertifi cate Send180ForCallWaiting NumberOfActiveDialogs EnableDID EnableDID_X NTTDIDSignallingForm FarEndDisconnectSilen ceThreshold T38UseRTPPort Valid Range and Description 0 Remove the ptime header from SDP 1 Include the ptime header in SDP default 0 Sets the destination number to the user par
50. the Device Information screen is displayed Figure 5 59 Device Information Screen Device Information MAC Address 0090808499 Serial Number 544665 Board Type 56 Device Up Time Od 17h 59m 4s 75th Device Administrative State Unlocked Device Operational State Enabled Flash Size bytes 6398608 RAM Size bytes 33554432 CPU Speed MHz 40 Version ID 5 004 010 DSP Type 0 DSP Software Version 20912 DSP Software Name 2041M Flash Version 195 Loaded Files Call Progress Tones File Name usa_tones_12 dat FXS Coefficient File Name MP11x 02 1 FXS_16KHZ dat Loaded Coder Table Default CODERTABLE gt To delete any of the loaded files take these 4 steps 1 Click the Delete button to the right of the files you want to delete Deleting a file takes effect only after the MediaPack is reset 2 Click the Maintenance button on the main menu bar the Maintenance Actions screen is displayed 3 In the Burn to FLASH field select Yes 4 Click the Reset button The gateway is reset and the files you chose to delete are removed Version 5 0 193 December 2006 a A wl AudioCodes MediaPack 5 7 4 Viewing the Ethernet Port Information The Ethernet Port Information screen provides read only information on the Ethernet connection used by the MediaPack The Ethernet Port Information parameters are displayed in Table 5 61 For
51. 1 167 Boot file name ram35136 cmp INI file name mp108 ini Call agent IP address 10 1 1 18 Log server IP address 0 0 0 0 Full Half Duplex state HALF DUPLEX Flash Software Burning state OFF Serial Debug Mode OFF Lan Debug Mode OFF BootLoad Version 1 75 Starting TFTP download Done MP108 Version 3 80 00 SIP User s Manual 262 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 3 Automatic Update Mechanism The MediaPack is capable of automatically updating its cmp ini and configuration files These files can be stored on any standard Web FTP or NFS server s and can be loaded periodically to the gateway via HTTP HTTPS FTP or NFS This mechanism can be used even for Customer Premise s Equipment CPE devices that are installed behind NAT and firewalls The Automatic Update mechanism is applied separately to each file For the detailed list of available files and their corresponding parameters refer to Table 5 56 on page 184 The Automatic Update mechanism assumes the external Web server conforms to the HTTP standard If the Web server ignores the If Modified Since header or doesn t provide the current date and time during the HTTP 200 OK response the gateway may reset itself repeatedly To overcome this problem adjust the update frequency AutoUpdateFrequency Three methods are used to activate the Automatic Update mechanism m After the MediaPack starts up refe
52. 2 Configuring the Advanced Parameters Use this submenu to configure the gateway s advanced control protocol parameters 5 5 2 1 General Parameters Use this screen to configure general control protocol parameters SIP User s Manual 76 Document LTRT 65408 SIP User s Manual 5 Web Management gt To configure the general parameters under Advanced Parameters take these 4 steps 1 Open the General Parameters screen Protocol Management menu gt Advanced Parameters submenu gt General Parameters option the General Parameters screen is displayed Figure 5 9 Advanced Parameters General Parameters Screen General Parameters IP Security Disable Filter Calls to IP Dont Filter Enable Digit Delivery to Tel Disable Enable Digit Delivery to IP Disable RTP Only Mode Disable Enable DID Wink Disable Delay Before DID Wink 0 Reanswer Time Disconnect and Answer Supervision 0 Enable Polarity Reversal Disable Enable Current Disconnect Disable Disconnect on Broken Connection Yes Broken Connection Timeout 100 msec 100 Disconnect Call on Silence Detection No Silence Detection Period sec 120 Silence Detection Method Voice Energy Detectors Send Digit Pattern on Connect CDR and Debug CDR Server IP Address CD
53. 209 gt To configure the date and time of the MediaPack take these 3 steps 1 Open the Regional Settings screen Advanced Configuration menu gt Regional Settings the Regional Settings screen is displayed Figure 5 46 Regional Settings Screen Regional Settings Send Call Progress Tones file fram your computer to the device Browse _ Send File Send FXS Coefficient file fram your computer to the device Browse _ Send File Send Voice Prompts file from your computer to the device eT Browse _ Send File DD Hour Min Sec 2 Enter the time and date where the gateway is installed 3 Click the Set Date amp Time button the date and time are automatically updated Note After performing a hardware reset the date and time are returned to their defaults and should be updated SIP User s Manual 166 Document LTRT 65408 SIP User s Manual 5 Web Management gt To load a configuration file to the gateway take these 8 steps 1 Open the Regional Settings screen Advanced Configuration menu gt Regional Settings the Regional Settings screen is displayed shown in Figure 5 46 2 Click the Browse button adjacent to the file you want to load 3 Navigate to the folder that contains the file you want to load 4 Click the file and click the Open button the name and path of the file appear in the field beside the Browse button
54. 217 December 2006 7a Ta P wt AudioCodes MediaPack 8 1 6 8 2 Message Waiting Indication Support for Message Waiting Indication MWI according to IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to MWI server MediaPack FXS gateways 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 Section 15 1 on page 325 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 gateway can subscribe to the MWI server per port usually used on FXS or per gateway used on FXO To configure MWI set the following parameters EnableMWI MW1ServerlP MW1AnalogLamp MW IDisplay StutterToneDuration EnableMW1Subscription MWIExpirationTime SubscribeRetryTime SubscriptionMode CallerlDType determines the standard for detection of MWI signals ETSIVMWITypeOneStandard Bellcore VMWITypeOneStandard Configuring the DTMF Transport Types You can control the way DTMF digits are transported over the IP network to the remote endpoint The following five modes are supported 1 Using INFO message according to the Nortel IETF draft In this mode DTMF digits are carried to the remote side within INFO messages To enable this m
55. 255 0 0 0 0 65535 Any 0 40000 50000 BLOCK 0 2 e fio 31 AD 255 255 255 0 4000 feooo Any fo 0 0 Block x 0 a 0 a B 10 4 0 0 255 255 0 0 4000 9000 Any o BLOCK 2 In the New Rule Index field enter the index of the access rule that you want to add 3 Click the Add an Empty Rule button a new rule appears alternatively click the Copy Selected Rule as a New Rule button a new rule that is an exact copy of the currently selected rule appears 4 Configure the rule s parameters according to Table 5 49 5 Click one of the following buttons e Apply Rule Settings to save the new rule the rule isn t active e Activate Rule to save the new rule and activate it e Delete Rule to delete the rule 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 gt To edit a rule take these 5 steps 1 Select the radio button of the entry you want to edit 2 Click the Make Rule Editable button the rule s fields can now be modified 3 Modify the fields according to your requirements 4 Click the Apply Rule Settings button to save the changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 gt To activate a de activated rule take these 2 steps 1 Select the radio button of the entry you want to activate 2 Click the Activate Rule button the rule is active gt To de activate an activate rule take these 2 st
56. 5 12 Configuring Voice Mail VM Parameters Use this screen to configure the VM parameters The VM application applies only to FXO gateways For detailed information on VM refer to the CPE Configuration Guide for Voice Mail gt To configure the VM parameters take these 4 steps 1 Open the Voice Mail screen Protocol Management menu gt FXO Settings gt Voice Mail option the Voice Mail screen is displayed Figure 5 34 Voice Mail Screen Voice Mail General Voice Mail Interface None Line Transfer Mode Disable Digit Patterns Forward on Busy Digit Pattern Forward on No Answer Digit Pattern Forward on Do Not Disturb Digit Pattern Forward on No Reason Digit Pattern Internal Call Digit Pattern External Call Digit Pattern Disconnect Call Digit Pattern MWI Off Digit Pattern MWI On Digit Pattern Enable SMDI Disable SMDI Timeout msec 2000 2 Configure the Voice Mail parameters according to Table 5 34 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 34 Voice Mail Parameters continues on pages 130 to 131 Parameter Description General Voice Mail Interface Enables the voice mail VM application on the gateway and determines the VoiceMaillnterface communication
57. 8 1 1 2 Receiving Hold Retrieve When an active call receives REINVITE message with either the IP address 0 0 0 0 or the inactive string in SDP the gateway stops sending RTP and plays a local Held Tone When an active call receives REINVITE message with sendonly string in SDP the gateway 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 to be played over IP by the remote party Version 5 0 215 December 2006 Ta WH wt AudioCodes MediaPack 8 1 2 8 1 3 Consultation Alternate m The Consultation feature is relevant only for the holding party applicable only to the MediaPack FXS gateway m After holding a call by pressing hook flash the holding party hears dial tone and can now initiate a new call which is called a consultation call E 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 E The held call can t be retrieved while Ringback tone is heard m After the consultation call is connected the user can switch between the held and active call by pressing hook flash Call Transfer There are two types of call transfers E Consultation Transfer REFER REPLACES E Blind Transfer REFER The common way to perform a consultation transfer is as follows In the transfer scenario there are three parties Party
58. A transferring Party B transferred Party C transferred to A Calls B B answers A presses the hook flash and puts B on hold party B hears a hold tone A dials C After A completed dialing C he can perform the transfer by onhook the A phone After the transfer is completed 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 to C m Just after completing dialing C phone number Transfer from setup E While hearing Ringback Transfer from alert m While speaking to C Transfer from active Blind transfer is performed after we have a call between A and B and party A decides to transfer the call to C immediately without speaking with C The result of the transfer is a call between B and C just like consultation transfer only skipping the consultation stage Note the following SIP issues E Transfer is initiated by sending REFER with REPLACES m The gateway can receive and act upon receiving REFER with or without REPLACES m The gateway can receive and act upon receiving INVITE with REPLACES in which case the old call is replaced by the new one E The INVITE with REPLACES can be used to implement Directed Call Pickup SIP User s Manual 216 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 1 4 8 1 5 Call Forward Five forms of call forward are supported Immediate Any incoming call is forwarded immediately and
59. Accept Accept Encoding Alert Info Allow Also Asserted Identity Authorization Call ID Call Info Contact Content Disposition Content Encoding Content Length Content Type Cseq Date Diversion Encryption Expires Fax From History Info Join Max Forwards Messages Waiting MIN SE Organization P Asserted Identity P Preferred Identity Priority Proxy Authenticate Proxy Authorization Proxy Require Prack Reason Record Route Refer To Referred By Replaces Require Remote Party ID SIP Headers The following SIP Headers are supported by the gateway Table B 3 SIP Headers continues on pages 342 to 343 Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No Yes Yes Yes Yes Yes Yes Yes Yes No Yes Yes No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes SI P User s Manual 342 Document LTRT 65408 SIP User s Manual B SIP Compliance Tables Table B 3 SIP Headers continues on pages 342 to 343 Header Field Response Key Retry After Route Rseq Session Expires Server SIP If Match Subject Supported Timestamp To Unsupported User Agent Via Voicemail Warning WWW Authenticate B 4 SDP Headers The following SDP Headers are supported by the gateway Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Table B 4 SDP Headers SDP Header Element Supported
60. Advanced Example of an Access List Definition via ini File ACCESSLIST FORMAT AccessList_Index AccessList_Source_IP AccessList_Net_Mask AccessList_Start_Port AccessList_End_Port AccessList_Protocol AccessList_Packet_Size AccessList_Byte_Rate AccessList_Byte_Burst AccessList_Allow_Type Necesshiste IO 10 000 255 0 0 0 O SaSaa airy O 2OO00 FOOOO allllery AccesslList 5 OR sia Op 255 255 255 0 4000 9000m amy 0r 0 07 clown ACCES Sits tan Oe mrt OPO PO mm Ol Or OO mm Ol 5535 cia Ol Oa Ol ooe ACCESS MESI Explanation of the example access list This access list consists of three rules m Rule 10 traffic from the subnet 10 xxx yyy zzz is allowed if the traffic rate does not exceed 40 KB s m Rule 15 if a packet didn t match rule 10 that is the excess traffic is over 40 KB s and coming from the subnet 10 31 4 xxx to ports 4000 to 9000 then it is allowed Rule 20 all other traffic which didn t match the previous rules is blocked The internal firewall can also be configured via the Embedded Web Server refer to Section 5 6 5 3 on page 171 SIP User s Manual 298 Document LTRT 65408 SIP User s Manual 12 Security 12 6 Network Port Usage The following table lists the default TCP UDP network port numbers used by the MediaPack Where relevant the table lists the ini file parameters that control the port usage and provide source IP address filtering capabilities
61. BootP Tool so that no replies are sent to BootP if requests Click the button again to restart the BootP Tool so that it responds to all BootP requests The Pause button provides a depressed graphic when the feature is active Edit Clients Click this button to open a new window that enables you to enter configuration information for each supported VoIP gateway Details on the Clients window are provided in Section C 11 on page 355 Edit Templates Click this button to open a new window that enables you to create or edit standard templates These templates can be used when configuring new clients that share most of the same settings Details on the Templates window are provided in Section C 12 on page 360 Clear Log Click this button to clear all entries from the Log Window portion of the I main application screen Details on the log window are provided in Section C 9 on page 352 Filter Clients Click this button to prevent the BootP Tool from logging BootP requests received from disabled clients or from clients which do not have entries in the Clients table Reset Click this button to open a new window where you enter an IP address requests for a gateway that you want to reset Version 5 0 351 December 2006 a WH wt AudioCodes MediaPack C 9 Figure C 2 Reset Screen IP to Reset t Ba Close When a gateway resets it first sends a BootRequest Therefore Reset can be used to force a BootP session wi
62. D where X is any integer between 0 and 4 0 sets the first table entry 1 sets the second and so on and D is an integer between 0 and 255 14 8 3 2 Configuration of Trusted Managers via SNMP To configure Trusted Managers the EM must use the SNMP COMMUNITY MIB the snmpTargetMIB and the TGT ADDRESS MASK MIB gt To add the first Trusted Manager take these 3 steps The following procedure assumes that there is at least one configured read write community There are currently no Trusted Managers The taglist for columns for all srCommunityTable rows are currently empty 1 Add a row to the snmpTargetAddrTable with these values Name mgr0 TagList MGR Params v2cparams 2 Add a row to the tgtAddressMaskTable table with these values Name mgr0 tgtAddressMask 255 255 255 255 0 The agent doesn t allow creation of a row in this table unless a corresponding row exists in the snmpTargetAddrTable 3 Set the value of the TransportLabel field on each non TrapGroup row in the srCommunityTable to MGR The following procedure assumes that there is at least one configured read write community There are currently one or more Trusted Managers The taglist for columns for all rows in the srCommunityTable are currently set to MGR This procedure must be performed from one of the existing Trusted Managers gt To adda subsequent Trusted Manager take these 2 steps 1 Add a row to the snmpTargetAddrTable with these values Name mgrN Ta
63. Default Coder Group v 2 From the Profile ID drop down list select the IP Profile you want to edit up to four IP Profiles can be configured 3 In the Profile Name field enter a name that enables you to identify the Profile intuitively and easily Version 5 0 113 December 2006 Ta WH wt AudioCodes MediaPack Parameter IPProfile_ID From the Profile Preference drop down list select the preference 1 20 of the current Profile The preference option is used to determine the priority of the Profile Where 20 is the highest preference value 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 IPProfile_ID 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 an intersection of the coders is performed i e only common coders remain 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 the description of the screen in which it is configured as an individual parameter From the Coder Group drop down list select the coder group you want to assign to that Profile You can select
64. E 4 opens Figure E 4 Prerecorded Tones Screen Prerecorded Tones File s r Prerecorded Tones File s Add files by dropping or using the Add File s button Add File s ToneType Name Coe gt 0 fiefs Remove Remove all Output C prerecordedtones dat Progress SIP User s Manual 366 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools 4 To add the prerecorded tone files you created in Step 1 to the Prerecorded Tones screen follow one of these procedures e Select the files and drag them to the Prerecorded Tones screen e Click the Add File s button the Select Files screen opens Select the required Prerecorded Tone files and click the Add gt gt button Close the Select Files screen 5 For each raw data file define a Tone Type a Coder and a Default Duration by completing the following steps e Double click or right click the required file the File Data window shown in Figure E 5 appears e From the Type drop down list select the tone type this raw data file is associated with e From the Coder drop down list select the coder that corresponds to the coder this raw data file was originally recorded with e Inthe Description field enter additional identifying information optional e Inthe Default field enter the default duration this raw data file is repeatedly played e Close
65. Endpoint 0 or Per Hunt Group 1 If not configured 1 the value of AuthenticationMode is used For example TrunkGroupSettings 1 5 Note This parameter can appear up to 24 times The gateway s port numbers are defined in the Endpoint Phone Numbers table under the Phone Number column For detailed information on the Endpoint Phone Numbers table refer to Section 5 5 7 on page 115 SIP User s Manual 118 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 9 5 5 9 1 Configuring the Endpoint Settings The Endpoint Settings screens enable you to configure port specific parameters Authentication The Authentication Table normally used with FXS gateways defines a username and password combination for authentication for each MediaPack port The Authentication Mode parameter described in Table 5 4 determines if authentication is performed per port or for the entire gateway If authentication is performed for the entire gateway this table is ignored Note If either the username or password field is omitted the port s phone number defined in Table 5 25 and global password refer to the parameter Password described in Table 5 4 are used instead gt To configure the Authentication Table take these 6 steps 1 Set the Authentication Mode parameter to Authentication per Endpoint 2 Open the Authentication screen Protocol Management me
66. Hostname gt lt first IP address gt lt second IP address gt For example DNS2IP Domainname com 10 8 21 4 10 13 2 95 Note This parameter can appear up to 10 times Version 5 0 105 December 2006 7a VT r wt AudioCodes MediaPack 5 5 5 5 Internal SRV Table The Internal SRV table is used for resolving host names to DNS A Records Three different A Records can be assigned to a hostname Each A Record contains the host name priority weight and port Note If the Internal SRV table is configured the gateway first tries to resolve a domain name using this table If the domain name isn t found the gateway performs an SRV resolution using an external DNS server To configure the Internal SRV table take these 9 steps Open the Internal SRV Table screen Protocol Management menu gt Routing Tables submenu gt Internal SRV Table option the Internal SRV Table screen is displayed Figure 5 20 Internal SRV Table Screen Internal SRV Table a JIL I IIL L L StL OO AEAII 1 F 3 4 5 6 7 8 g oa i e a 2 In the Domain Name field enter the hostname to be translated You can enter a string up to 31 characters long 3 From the Transport Type drop down list select a transport type 4 In the DNS Name 1 field enter the first DNS A Record to which the hostname is translated 5 I
67. IP address of the FXO MediaPack gateway 10 1 10 2 in the field IP Address Note In remote extensions for the transfer to function hold must be disabled on the FXS i e Enable Hold 0 8 12 4 4 FXO MediaPack Configuration using the Embedded Web Server gt To configure the FXO MediaPack take these 4 steps 1 In the Endpoint Phone Numbers screen assign the phone numbers 200 to 207 for the gateway s endpoints Endpoint Phone Number Table 1 FXO 1 8 200 0 2 In the Automatic Dialing screen enter the phone numbers of the FXS MediaPack gateway in the Destination Phone Number fields When a ringing signal is detected at port 1 the FXO gateway automatically dials the number 100 Automatic Dialing Gateway Port Destination Phone Number Auto Dial Status Port 1 FXO 100 Enable Port 2 FXO 101 Enable Port 3 FXO 102 Enable Port 4 FxO 103 Enable Port FXO 104 Enable Port6 FXO 105 Enable Port FXO 106 Enable Port 8 FXO Enable 3 In the Tel to IP Routing screen enter 10 in the Destination Phone Prefix field and the IP address of the FXS MediaPack gateway 10 1 10 3 in the field IP Address 4 In the Protocol Management screen set the parameter Dialin
68. If the Hold service is enabled a user can activate Hold or Unhold using the hook flash On receiving a Hold request the remote party is put on hold and hears the hold tone Note To use this service the gateways at both ends must support this option Determines the format of the hold request 0 0 0 0 0 The connection IP address in SDP is 0 0 0 0 default Send Only 1 The last attribute of the SDP contains the following a sendonly No 0 Disable the call transfer service Yes 1 Enable the call transfer service using REFER default If the 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 Note 1 To use this service the gateways at both ends must support this option Note 2 To use this service set the parameter Enable Hold to Yes Defined string that is added as a prefix to the transferred forwarded called number when REFER 3xx message is received Note 1 The number manipulation rules apply to the user part of the REFER TO Contact URI before it is sent in the INVITE message Note 2 The xferprefix parameter can be used to apply different manipulation rules to differentiate the transferred forwarded number from the original dialed number No 0 Disable the Call Forward service Yes 1 Enable Call Forward service using REFER default For FXS gateways a
69. IsFaxUsed is other than 1 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 Initiate T 38 on Preamble 0 Terminating fax gateway initiates T 38 session on receiving of HDLC preamble signal from fax default Initiate T 38 on CED 1 Terminating fax gateway initiates T 38 session on receiving of CED answer tone from fax Note This parameters is applicable only if IsFaxUsed 1 Determines the default transport layer used for outgoing SIP calls initiated by the gateway UDP 0 default TCP 1 TLS 2 SIPS Note It is recommended to use TLS to communicate with a SIP Proxy and not for direct gateway gateway communication Version 5 0 59 December 2006 Ta WH wt AudioCodes MediaPack Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter SIP UDP Local Port LocalSIPPort SIP TCP Local Port TCPLocalSIPPort SIP TLS Local Port TLSLocalSIPPort Enable SIPS EnableSIPS Enable TCP Connection Reuse EnableTCPConnectionRe use SIP Destination Port SIPDestinationPort Use user phone in SIP URL IsUserPhone Use user phone in From Header IsUserPhonelnFrom Use Tel URI for Asserted Identity UseTelURIForAssertedID Tel to IP No Answer Timeout IPAlertTimeout Enable Remote Party ID EnableRPlheader Description Local
70. MediaPack Table F 2 acBoardTemperatureAlarm Alarm Trap Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status Condition Alarm status Corrective Action Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status lt text value gt Condition Alarm status lt text value gt Condition Alarm status Corrective Action acBoardTemperatureAlarm 1 3 6 1 4 1 5003 9 10 1 21 2 0 3 Critical equipmentAlarm temperatureUnacceptable 50 Board temperature too high Temperature is above 60 degrees C 140 degrees F Critical After raise temperature falls below 55 degrees C 131 degrees F Cleared Inspect the system Determine if all fans in the system are properly operating Table F 3 acgwAdminStateChange Alarm Trap acgwAdminStateChange 1 3 6 1 4 1 5003 9 10 1 21 2 0 7 Major processingErrorAlarm outOfService 71 Network element admin state change alarm Gateway is lt text gt Admin state changed to shutting down Major Shutting down No time limit Admin state changed to locked Major Locked Admin state changed to unlocked Cleared A network administrator has taken an action to lock the device No corrective action is required SIP User s Manual 376 Document LTRT 65408 SIP User s Manual F SNMP Traps Alarm OID Default Severity
71. Number Plan SrcPhoneNum Source Phone Number SrcNumBeforeMap Source Number Before Manipulation TON Destination Phone Number Type NPI Destination Phone Number Plan DstPhoneNum Destination Phone Number DstNumBeforeMap Destination Number Before Manipulation Durat Call Duration Coder Selected Coder Intrv Packet Interval Rtplp RTP IP Address SIP User s Manual 232 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities Table 8 1 Supported CDR Fields continues on pages 232 to 233 Field Name Port TrmSd TrmReason Fax InPackets OutPackets PackLoss Uniqueld SetupTime ConnectTime ReleaseTime RTPdelay RTPjitter RTPssrc RemoteRTPssrc RedirectReason TON NPI RedirectPhonNum Description 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 Number of Incoming Lost Packets unique RTP ID Call Setup Time Call Connect Time Call Release Time RTP Delay RTP Jitter Local RTP SSRC Remote RTP SSRC Redirect Reason Redirection Phone Number Type Redirection Phone Number Plan Redirection Phone Number 8 10 Supported RADIUS Attributes Use Table 8 2 below for explanations on the RADIUS attributes contained in the communication packets transmitted between the MediaPack and a RADIUS Server Table 8 2 Supported RADIUS Attributes continues on pages 233 to 235 Attribute Attri
72. 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 1 and not exceeding the number of Call Progress Tones defined in the first section using the following keys Tone Type Call Progress Tone type Figure 15 1 Call Progress Tone Types O OO W NS Dial Tone Ringback Tone Busy Tone Reorder Tone Confirmation Tone Call Waiting Tone 15 Stutter Dial Tone 16 Off Hook Warning Tone 17 Call Waiting Ringback Tone 23 Hold Tone Tone Modulation Type Either Amplitude Modulated 1 or regular 0 Tone Form The tone s format can be one of the following Continuous Cadence Burst Low Freq Hz Frequency in hertz of the lower tone component in case of dual frequency tone or the frequency of the tone in case of single tone not relevant to AM tones High Freq Hz Frequency in hertz 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 0 to 31 dBm The value should be set to 32 in the case of a single tone not relevant to AM tones First Signal On Time 10 msec Signal On
73. RAI Resource Available Indication service default 1 Enable RAI service If RAI is enabled an SNMP acBoardCallResourcesAlarm Alarm Trap is sent if gateway resources fall below a predefined configurable threshold High Threshold in percentage that defines the gateway s busy endpoints The range is 0 to 100 The default value is 90 When the percentage of the gateway s busy endpoints exceeds the value configured in High Threshold the gateway sends an SNMP acBoardCallResourcesAlarm Alarm Trap with a major Alarm Status Note The gateway s available Resources are calculated by dividing the number of busy endpoints by the total number of available gateway endpoints Low Threshold in percentage that defines the gateway s busy endpoints The range is 0 to 100 The default value is 90 When the percentage of the gateway s busy endpoints falls below the value defined in Low Threshold the gateway sends an SNMP acBoardCallResourcesAlarm Alarm Trap with a cleared Alarm Status Time interval in seconds that the gateway checks for resource availability The default is 10 seconds SIP User s Manual 134 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 35 Protocol Management ini File Parameters continues on pages 132 to 137 ini File Parameter Name 3WayConferenceMode WarningToneDuration FXONumberOfRings PhoneContext Valid Range and Descriptio
74. RTP timestamp 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 Version 5 0 159 December 2006 Ta Ao wi AudioCodes MediaPack 5 6 2 4 Configuring the Hook Flash Settings gt To configure the Hook Flash Settings parameters take these 4 steps Open the Hook Flash Settings screen Advanced Configuration menu gt Media Settings gt Hook Flash Settings option the Hook Flash Settings screen is displayed Figure 5 43 Hook Flash Settings Screen Hook Flash Settings Min Hook Flash Detection Period msec 100 Hook Flash Detection Period msec 700 2 Configure the Hook Flash Settings according to Table 5 45 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 45 Media Settings Hook Flash Settings Parameters Parameter Description Min Flash Hook Detection Period Sets the minimal time in msec for detection of a flash hook event for msec FXS only MinFlashHookTime The valid range is 25 to 300 The default value is 300 msec Detection is guaranteed for flash hook periods of at least 60 msec when setting the minimal time to 25 Flash hook signals that last a shorter period o
75. RoutingTableGatewaysColumn destination matches the rules in the adjacent columns A read only field that indicates the time period for which the specific TTL i aa KOR GN routing rule is valid The lifetime of a static route is infinite Hop Count The maximum number of allowed routers between the gateway and RoutingTableHopsCountColumn destination Network Type Specifies the network type the routing rule is applied to RoutingTablelnterfacesColumn OAM 0 default Control 1 Media 2 For detailed information on the network types refer to Section 9 8 on page 253 ini File Example The IP routing ini file 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 forth In the following example two rows are configured when the gateway 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 0 1 Routing TableHopsCountColumn 20 20 SIP User s Manual 146 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 1 5 Configuring the VLAN Settings For detailed information on the MediaPack VLAN implementation refer to Section 9 8 on page 253 gt To configure the V
76. Settings gt General Security Settings option the General Security Settings screen is displayed Figure 5 50 General Security Settings Screen General Security Settings Secured Web Connection HTTPS HTTP and HTTPS HTTP Authentication Mode Digest when possible Voice Menu Password General RADIUS Settings Enable RADIUS Access Control Disable Use RADIUS for Web Telnet Login j gt _ _ _ v Disable RADIUS Authentication Server IP Address 0 0 0 0 RADIUS Authentication Server Port 1645 RADIUS Shared Secret RADIUS Authentication Settings Default Access Level 200 Device Behavior Upon RADIUS Timeout Verify Access Locally Local RADIUS Password Cache Mode Reset Timer Upon Access Local RADIUS Password Cache Timeout sec 300 RADIUS YSA Vendor ID 5003 RADIUS VSA Access Level Attribute EtherDiscover Settings 36 EtherDiscover Operation Mode SRIP Settings Enable if unconfiqured Enable Media Security Disable Media Security Behavior IPSec Settings Preferable Enable IP Security Disable 2 Configure the General Security Settings according to Table 5 50 below 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer t
77. To decode an encoded ini file take these 4 steps Click the Select File button under the Decode ini File s section Navigate to the folder that contains the file you want to decode Click the file and click the Open button the name and path of both the encode ini file and the output decoded file appear in the fields under the Select File button Note that the name of the output file can be modified Click the Decode File s button a decoded ini file with the name you specified is created Note that the decoding process verifies the input file for validity Any change made to the encoded file causes an error and the decoding process is aborted Version 5 0 365 December 2006 7a Ta ie wt AudioCodes MediaPack E 1 3 Creating a Loadable Prerecorded Tones File For detailed information on the PRT file refer to Section 15 2 on page 330 Note The maximum size of a PRT file that can be loaded to the gateway is 100 KB gt To create a loadable PRT dat file from your raw data files take these 7 steps 1 Prepare the prerecorded tones raw data PCM or L8 files you want to combine into a single dat file using standard recording utilities 2 Execute the TrunkPack Downloadable Conversion utility DConvert exe supplied with the software package the utility s main screen opens shown in Figure E 1 3 Click the Process Prerecorded Tones File s button the Prerecorded Tones File s screen shown in Figure
78. a media server or other off device system The device provides two types of performance measurements m Gauges Gauges represent the current state of activities on the device Gauges unlike counters can decrease in value and like counters can increase The value of a gauge is the current value or a snapshot of the current activity on the device Counters Counters always increase in value and are cumulative Counters unlike gauges never decrease in value unless the off device system is reset the counters are then zeroed Performance measurements are provided by several proprietary MIBs that are located under the performance sub tree iso 1 org 3 dod 6 internet 1 private 4 enterprises 1 audioCodes 5003 acPerformance 10 Two formats of performance monitoring MIBs are available Old format obsolete as of version 4 6 Each MIB is composed of a list of single MIB objects each relates to a separate attribute within a gauge or a counter All counters and gauges provide the current time value only e acPerfMediaGateway a generic type of PM MIB that covers Control protocol RTP stream System packets statistics e acPerfMediaServices Media services devices specific performance MIB e acPerfH323SIPGateway holds statistics on Tel to IP and vice versa m New format The following MIBs feature an identical structure Each includes two major sub trees e Configuration sub tree enables configurat
79. a semicolon E End of Table Mark Indicates the end of the table The same string used for the table s title preceded by a backslash e g IAMY_TABLE_NAME Figure 10 4 displays an example of the structure of a parameter table in the ini file Figure 10 4 Structure of a Parameter Table in the ini File Table Items Table Fields Item_Name Item_Serial_Number Item_Color Item_weight NOTE Item_Color is not specified It will be given default value Items_Table Fields declaration Format Item_Index Item_Name Item_Serial_Number Item_weight tems_Table 0 Computer 678678 6 tems_Table 6 Computer screen 127979 9 tems_Table 2 Computer pad 111111 Items_Table Version 5 0 269 December 2006 Ta WH wt AudioCodes MediaPack Refer to the following notes Indices in both the Format and the Data lines must appear in the same order determined by the specific table s documentation The Index field must never be omitted The Format line can include a sub set 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 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 The sign in a Data line indicates that the user wants to assign the pre defin
80. and the identifier traffic type of the VLAN to which it belongs media control or management IEEE 802 1Q The class of service mechanism can be utilized to accomplish Ethernet QoS Packets sent by the MediaPack to the Ethernet network are divided into five different priority classes Network Premium media Premium control Gold and Bronze The priority of each class is determined by a corresponding ini file parameter Traffic type tagging can be used to implement Layer 2 VLAN security By discriminating traffic into separate and independent domains the information is preserved within the VLAN Incoming packets received from an incorrect VLAN are discarded SIP User s Manual 254 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities For the mapping of an application to its class of service and traffic type refer to Table 9 1 below Media traffic type is assigned Premium media class of service Management traffic type is assigned Bronze class of service and Control traffic type is assigned Premium control class of service For example RTP RTCP traffic is assigned the Media VLAN ID and Premium media class of service whereas Web traffic is assigned the Management VLAN ID and Bronze class of service Each of these parameters can be configured with a 802 1p q value traffic type to VLAN ID and class of service to 802 1p priority As a safety measure the VLAN mechanism is acti
81. and username retain their values default 1 Password and username are reset for the default username and password refer to Table 4 1 on page 41 Note The username and password cannot be reset from the Web i e via AdminPage or by loading an ini file 0 Enable changing parameters from Web default 1 Operate Web server in read only mode HTTP port used for Web management default 80 Destination IP address in dotted format notation to which the gateway sends proprietary UDP ping packets The default IP address is 0 0 0 0 Destination UDP port to which the heartbeat packets are sent The range is 0 to 64000 The default is 0 Delay in msec between consecutive heartbeat packets 10 100000 1 disabled default SIP User s Manual 182 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 55 Board ini File Parameters continues on pages 182 to 184 ini File Parameter Name RADIUSRetransmission RADIUSTo HTTPS Parameters HTTPSPort HTTPSRequireClientCertificate HTTPSRootFileName HTTPSPkeyFileName Security Settings gt Certificates HTTPSCertFileName BootP and TFTP Parameters Valid Range and Description Determines the number of RADIUS retransmission retries for the same request The valid range is 1 to 10 The default value is 3 Determines the time interval measured in seconds the gateway waits for a response before a RADIUS retransmissio
82. appears when the save is completed successfully Version 5 0 205 December 2006 r wi AudioCodes MediaPack 5 10 3 Resetting the MediaPack The Maintenance Actions screen enables you to remotely reset the gateway Before you reset the gateway you can choose the following options Save the gateway s current configuration to the flash memory non volatile Perform a graceful shutdown Reset starts only after a user defined time expires or no more active traffic exists the earliest thereof To reset the gateway take these 5 steps Open the Maintenance Actions screen Maintenance menu the Maintenance Actions screen is displayed Figure 5 73 Maintenance Actions Screen Maintenance Actions Reset Board Burn To FLASH Graceful Option Shutdown Timeout sec LOCK UNLOCK Lock Graceful Option Current Admin State UNLOCKED Save Configuration Save Configuration Under the RESET group from the Burn To FLASH drop down list select one of the following options e Yes The gateway s current configuration is burned i e saved to the flash memory prior to reset default e _ No Resets the device without burning i e saving the current configuration to flash discards all unsaved modifications to the configuration Under the RESET group from the Graceful Option drop down list select one of the following op
83. are combined into a single RTP packet The ptime specifies the packetization time the gateway expects to receive The gateway always uses the ptime requested by the remote side for sending RTP packets 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 for the coder you selected is dynamic enter a value from 0 to 120 payload types of well known coders cannot be modified The payload type identifies the format of the RTP payload From the Silence Suppression drop down list enable or disable the silence suppression option for the coder you selected Repeat steps 3 to 7 for the second to fifth coders optional Repeat steps 2 to 8 for the second to fourth coder groups optional Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Each coder can appear only once If not specified the ptime gets a default value The ptime specifies the packetization time the gateway expects to receive The gateway always uses the ptime requested by the remote side for sending RTP packets If payload type is not specified a default is used For G 729 it is also possible to select silence suppression without adaptations Only the ptime of the first coder in the defined coder list is declared in INVITE 200 OK SDP even if multiple coders are defined If
84. are designed and tested to be fully interoperable with leading softswitches and SIP servers The MediaPack gateways incorporate up to 24 analog ports for connection directly to an enterprise PBX FXO or and to phones fax machines and modems FXS supporting up to 24 simultaneous VolP calls Additionally the MediaPack units are equipped with a 10 100 Base TX Ethernet port for connection to the network The MediaPack gateways are best suited for small to medium size enterprises branch offices or for residential media gateway solutions The MediaPack gateways enable users to make free local or international telephone fax calls between the distributed company offices using their existing telephones fax These calls are routed over the existing network ensuring that voice traffic uses minimum bandwidth The MediaPack gateways are very compact devices that can be installed as a desk top unit on the wall or in a 19 inch rack The MediaPack gateways support SIP Session Initiation Protocol protocol enabling the deployment of voice over IP solutions in environments where each enterprise or residential location is provided with a simple media gateway This provides the enterprise with a telephone connection e g RJ 11 and the capability to transmit the voice and telephony signals over a packet network Version 5 0 19 December 2006 Ta A wi AudioCodes MediaPack The layout diagram Figure 1 1 illustrates a
85. by the answering side Some SIP gateways don t support the detection of this fax signal on the answering side thus for these cases it is possible to configure the gateways to start the T 38 fax session when the CNG tone is detected by the originating side However this mode is not recommended SIP User s Manual 156 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 2 3 Configuring the RTP RTCP Settings gt To configure the RTP RTCP Settings parameters take these 4 steps 1 Open the RTP RTCP Settings screen Advanced Configuration menu gt Media Settings gt RTP RTCP Settings option the RTP RTCP Settings screen is displayed Figure 5 42 RTP RTCP Settings Screen RTP RTCP Settings Dynamic Jitter Buffer Minimum Delay Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Packing Factor Basic RTP Packet Interval Default RTP Directional Control Transmit Receive RFC 2633 TX Payload Type 96 RFC 2633 RX Payload Type 96 RFC 2198 Payload Type Fax Bypass Payload Type Enable RFC 3369 CN Payload Type Disable Analog Signal Transport Type Ignore analog signals RTP Base UDP Port 6000 Remote RTP Base UDP Port RTP Multiplexing Local UDP Port IRTP Multiplexing Remote UDP Port
86. call The valid range is 0 to 255 The default is 0 seconds Defines the Phone Context table When a call is received from the ISDN the NPI and TON are compared against the table and the 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 PhoneContext lt Number Plan gt lt Number Type gt lt Phone Context gt For example PhoneContext 0 0 unknown com PhoneContext 1 1 host com PhoneContext 9 1 na e164 host com Note 1 This parameter can appear up to 20 times Note 2 Several rows with the same NPI TON or Phone Context are allowed In this scenario a Tel to IP call uses the first match Note 3 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 Note 4 To configure Phone Context table using the Web interface refer to Section 5 5 4 on page 96 Version 5 0 135 December 2006 Ta WH wt AudioCodes MediaPack Table 5 35 Protocol Management ini File Parameters continues on pages 132 to 137 ini File Parameter Name EnableRport Valid Range and Description Enables disables the usage of the rport
87. coder G 729 is selected and silence suppression disabled the gateway 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 5 0 109 December 2006 Ta WH wt AudioCodes MediaPack Parameter CoderName_ID Table 5 22 ini File Coder Group Parameter Description Defines groups of coders that can be associated with IP or Tel profiles up to five coders in each group Enter coder groups in the following format CoderName_ lt coder group ID from 1 to 4 gt lt Coder Name gt lt Ptime gt lt Rate gt lt Payload Type gt lt Silence Suppression Mode gt Note 1 This parameter CoderName_ID can appear up to 20 times five coders in four coder groups Note 2 The coder name is case sensitive Note 3 If silence suppression is not defined for a specific coder the value defined by the parameter EnableSilenceCompression is used Note 4 The value of several fields is hard coded according to well known standards e g the 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 For example CoderName_1 g 11Alaw64k 20 0 CoderNa
88. continues on pages 145 to 146 Column Name Description ini File Parameter Name Delete Row To delete IP routing rules from the IP Routing Table check the Delete Row checkbox in the rows of the routing rules you want to delete and click the button Delete Selected Entries the routing rules are removed from the table Destination IP Address RoutingTableDestinationsColum Specifies the IP address of the destination host network n Destination Mask RoutingTableDestinationMasksC Specifies the subnet mask of the destination host network olumn Version 5 0 145 December 2006 Ta WH wt AudioCodes MediaPack Table 5 39 IP Routing Table Column Description continues on pages 145 to 146 Column Name Description ini File Parameter Name The address of the host network you want to reach is determined by an AND operation that is applied on 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 Specifies the IP address of the router to which the packets are sent if their
89. default and must be activated using the parameter EnableUserlnfoUsage described in Section 5 5 2 1 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 Table 15 1 below An example of a User Information file is shown in Figure 15 5 below 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 i e initially the gateway 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 username and password are used for SIP Authentication when required 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 The User Information file is a text file the file size mustn t exceed 10 800 bytes that can be loaded via the ini file User
90. descending again Descending 4 Select the highest available channel Always start at the highest channel number in the hunt group and if that channel is not available select the next lower channel Dest Number Cyclic Ascending 5 First select the gateway port according to the called number called number is defined in the Endpoint Phone Number table If the called number isn t found then select 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 By Source Phone Number 6 Select the gateway port according to the calling number The default method is By Phone Number Enable Early Media Disable 0 Early Media is disabled default EnableEarlyMedia Enable 1 Enable Early Media If enabled the gateway sends 183 Session Progress response with SDP instead of 180 Ringing allowing the media stream to be set up prior to the answering of the call Note that to send 183 response you must also set the parameter ProgressIindicator2IP to 1 If it is equal to 0 180 Ringing response is sent Note Generally this parameter is set to 1 Session Expires Time Determines the timeout in seconds for keeping a re INVITE message alive within SIPSessionExpires a SIP session The SIP session is refreshed each time this timer expires The SIP method used for session timer updates is determined acco
91. enabled the gateway 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 Note The NAT mechanism and the IP address translation must be enabled for this parameter to take effect DisableNAT 0 EnablelpAddrTranslation 1 Enables or disables the transmission of RTP or T 38 No Op packets Valid options include 0 Disable default 1 Enable This mechanism ensures that the NAT binding remains open during RTP or T 38 silence periods This parameter is now obsolete Please use the parameter NoOperationSendingMode SIP User s Manual 150 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 41 Network Settings ini File Parameters continues on pages 149 to 151 ini File Parameter Name NoOplnierval RTPNoOpinterval RTPNoOpPayloadType EnableDetectRemoteMACChan ge Valid Range and Description 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 NoOperationSendingMode parameter This parameter is now obsolete Please use
92. equal to the endpoint phone number 122 The realm return by the proxy audiocodes com The password from the ini file AudioCodes The equation to be evaluated according to RFC this part is called A1 122 audiocodes com AudioCodes The MD5 algorithm is run on this equation and stored for future usage The result is a8f17d4b41ab8dab6c95d3c14e34a9e1 Next we need to evaluate the par called A2 We take The method type REGISTER Using SIP protocol sip Proxy IP from ini file 10 2 2 222 The equation to be evaluated REGISTER sip 10 2 2 222 The MD5 algorithm is run on this equation and stored for future usage The result is a9a031cfddcb10d91c8e7b4926086f7e The final stage The A1 result The nonce from the proxy response 11432d6bce58ddf02e3b5e1c77c010d2 The A2 result The equation to be evaluated A1 11432d6bce58ddf02e3b5e1c77c010d2 A2 The MD5 algorithm is run on this equation The outcome of the calculation is the response needed by the gateway to be able to register with the Proxy The response is b9c45d0234a5abf5ddf5c704029b38cf Version 5 0 241 December 2006 Ta WH wt AudioCodes MediaPack At this time a new REGISTER request is issued with the response REGISTER sip710 2 2 222 6iPy2 0 Waa Sie 2 0 Une 10 1 1 200 From lt sip 122 10 1 1 200 gt tag 1c23940 To seios 22E i 21 200 Geul Tys SSA SAE il 206 Server Audiocodes Sip Gateway MP 1
93. field Parameter Format TrunkGroup_ lt Hunt Group ID gt lt Starting channel gt lt Ending channel gt lt Phone Number gt lt Tel Profile ID gt For example TrunkGroup_1 1 4 100 TrunkGroup_2 5 8 200 1 Note 1 The numbering of channels starts with 1 Note 2 Hunt Group ID can be set to any number in the range 1 to 99 Note 3 This parameter can appear up to 8 times for 8 port gateways and up to 24 times for MP 124 gateways Note 4 An optional Tel ProfilelD 1 to 4 can be applied to each group of channels This parameter is obsolete use instead TrunkGroup_x SIP User s Manual 116 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 8 Configuring the Hunt Group Settings The Hunt Group Settings Table is used to determine the method in which new calls are assigned to channels within each hunt group If such a rule doesn t exist for a specific hunt group the global rule defined by the Channel Select Mode parameter Protocol Definition gt General Parameters applies gt To configure the Hunt Group Settings table take these 8 steps 1 z 3 4 5 6 ri 5 z oO Open the Hunt Group Settings screen Protocol Management menu gt Hunt Group Settings the Hunt Group Settings screen is displayed Figure 5 26 Hunt Group Settings screen Hunt Group ID Channel Select Mode Registration Mode 1 Cyclic Ascending Per Endpoint
94. first incoming packet To compare only the IP address set EnablelpAddrTranslation 1 and EnableUdpPortTranslation 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 No Op Packets The gateway 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 No Op packets are available for sending in RTP and T 38 formats Users can control the activation of No Op packets by using the ini file parameter NoOperationSendingMode If No Op packet transmission is activated users 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 NoOpinterval ini parameter Note Receipt of No Op is always supported RTP No OP The RTP No Op support complies with IETF s draft wing avt rtp noop 03 txt titled A No Op Payload Format for RTP This IETF document defines a No Op payload format for RTP The draft defines the RTP payload type as dynamic Users can control the payload type with which the No Op packets are sent This is performed using the RTPNoOpPayloadType ini parameter AudioCodes default payload type is 120 E T 38 No Op T 38 No Op packets are sent only while a T 38 s
95. gt vee w AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 5 0 Document LTRT 65408 December 2006 SIP User s Manual Contents Table of Contents BO EI S AE PAE AE A E AE A AE A he 1 1 Gateway Description ccs tease etch ccerdeoeeiseecm thes cuasiaia wpm tctesdereeaceeccenaeg 19 1 3 2 MediaPack Physical Description cscssssssssssssseessensesssenessnsnsessenssennes D 2 1 Metia T rama Desci pio ciarrniniotiminienee E 22 2 og MP 124 Rear P 3 In stalling the MediaPack SSSR RRR SEER EERE MES AETA AAE SEE A TNE NETE A E ot a1 instali ing ihg M P m IR E R A A E AET AET E E E E AE EEEE 4 3 Configure the MediaPack Basic Paramet E EE A E E E E E Version 5 0 3 December 20 06 7a wt AudioCodes MediaPack 5 Web PIAS cscs icsseds sonics pansies vneandnsvnnandnsvansndesnsnundpivenendparsonntntnsomtninneed 5 1 5 3 5 4 4 Searc ing jori 5 5 Protocol Managem Protocol D fin SIP User s Manual 4 Document LTRT 65408 SIP User s Manual Contents 147 Version 5 0 5 Decem ber 2006 7a wt AudioCodes MediaPack Using BootP SOT iss wseacdeasneccdineee nese tenriedennnn nsdn ternae fond enenintensaeieenmeeninernnantennns Ta eik endl Server Parameters 7 2 7 3 Telephony Ba N 15 8 1 g cine Su ae 8 2 gba iSite e DTMF Transport Types 8 3 dem 7 port Modes 0
96. includes the cmp file Determines the number of minutes the gateway waits between automatic updates The default value is 0 the update at fixed intervals mechanism is disabled Schedules an automatic update to a predefined time of the day The range is HH MM 24 hour format For example 20 18 Note The actual update time is randomized by five minutes to reduce the load on the Web servers Invokes an immediate restart of the gateway This option can be used to activate offline not on the fly parameters that are loaded via IniFileUrl 0 The immediate restart mechanism is disabled default 1 The gateway immediately restarts after an ini file with this parameter set to 1 is loaded Version 5 0 185 December 2006 7a Ta r wt AudioCodes MediaPack 5 6 6 6 SNMP jini File Parameters Table 5 57 describes the SNMP parameters that can only be configured via the ini file Table 5 57 SNMP ini File Parameters ini File Parameter Name Description SNMPPort The device s local UDP port used for SNMP Get Set commands The range is 100 to 3999 The default port is 161 SNMPTrustedMGR_x Up to five IP addresses of remote trusted SNMP managers from which the SNMP agent accepts and processes get and set requests Note 1 If no values are assigned to these parameters any manager can access the device Note 2 Trusted managers can work with all community strings SNMPManagerTrapUser_x This parameter can be set to the n
97. is 10 Note 1 Set to 13 for data fax amp modem calls Note 2 For more information on the Jitter Buffer refer to Section 8 6 on page 230 Enter 0 to disable the generation of redundant packets default Enter 1 to enable the generation of RFC 2198 redundancy packets N A Controlled internally by the gateway according to the selected coder N A Controlled internally by the gateway according to the selected coder N A Controlled internally by the gateway according to the selected coder N A Use the ini file parameter RFC2833PayloadType instead N A Use the ini file parameter RFC2833PayloadType instead RTP redundancy packet payload type according to RFC 2198 The range is 96 127 The default is 104 Applicable if RTP Redundancy Depth 1 Determines the fax bypass RTP dynamic payload type The valid range is 96 to 120 The default value is 102 Determines whether Silence Indicator SID packets that are sent and received are according to RFC 3389 Disable 0 G 711 SID packets are sent in a proprietary method default Enable 1 SID comfort noise packets are sent with the RTP SID payload type according to RFC 3389 Applicable to G 711 and G 726 coders Ignore analog signals 0 Hook flash isn t transferred to the remote side default RFC 2833 analog signal relay 1 Hook flash is transferred via RFC 2833 SIP User s Manual 158 Document LTRT 65408 SIP User s Manual 5 W
98. it detects the start of speech or ringback tone from the Tel side 8 4 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 gateway and only after receiving a dial tone from the PBX via the FXO gateway dials the destination telephone number Figure 8 2 Call Flow for Two Stage Dialing FXO Gateway SIP Client F1 INVITE FXO seizes line SIP User s Manual 224 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 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 will be The relevant parameters for configuring Dialing Time include the following DTMFDigitLength 100 msec time for generating DTMF tones to the PSTN PBX side DTMFinterDigitInterval 100 msec time between generated DTMF digits to PSTN PBX side 8 4 1 3 Call Termination Disconnect Supervision on the FXO Gateway The FXO Disconnect Supervision enables the gateway s FXO ports to monitor call progress tones from a PBX or from the PSTN This allows the FXO to determine when the call has terminated on the PBX side and thereby preventing analog trunks i e lines to the PBX from getting stuck when the called phone hangs up The PBX doesn t dis
99. message is displayed Version 5 0 55 December 2006 a WH wt AudioCodes MediaPack 5 5 Protocol Management Use this menu to configure the gateway s SIP parameters and tables Note Those parameters contained within square brackets are the names used to configure the parameters via the ini file 5 5 1 Protocol Definition Parameters Use this submenu to configure the gateway s specific SIP protocol parameters 5 5 1 1 General Parameters Use this screen to configure general SIP parameters SIP User s Manual 56 Document LTRT 65408 SIP User s Manual 5 Web Management gt To configure the general parameters under Protocol Definition take these 4 steps Open the General Parameters screen Protocol Management menu gt Protocol Definition submenu gt General Parameters option the General Parameters screen is displayed Figure 5 5 Protocol Definition General Parameters Screen PRACK Mode Supported iS Channel Select Mode Cyclic Ascending v Enable Early Media Disable j 183 Message Behavior Progress Session Expires Time 0 Minimum Session Expires 90 Session Expires Method Re Invite Ni Asserted Identi
100. messages If not specified the Subject header isn t included default The maximum length of the subject is limited to 50 characters Enables or disables the usage of the SIP Reason header Valid options include 0 Disable 1 Enable default Version 5 0 63 December 2006 Ta WH wt AudioCodes MediaPack Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter Enable Semi Attended Transfer EnableSemiAttendedTran sfer 3xx Behavior 3xxBehavior Multiple Packetization Time Format MultiPtimeFormat Description Determines the gateway behavior when Transfer is initiated while still in Alerting state Valid options include Disable 0 Send REFER with Replaces default Enable 1 Send CANCEL and after a 487 response is received send REFER without Replaces Determines the gateway s behavior when a 3xx response is received for an outgoing INVITE request The gateway can either use the same call identifiers CallID 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 Determines whether the mptime attribute is included in the outgoing SDP Valid options include 0 Disable default 1 Enable includes the mptime attribute in the outgoing SDP PacketCable defined format The mptime attribut
101. negotiation of the cipher suite fails the call is terminated Incoming calls that don t include encryption information are rejected default Enables disables the Secure Internet Protocol IPSec on the gateway Disable 0 IPSec is disabled default Enable 1 IPSec is enabled 5 6 5 6 Configuring the IPSec Table Use the IPSec Table screen to configure the IPSec parameters For detailed information on IPSec and IKE refer to Section 12 1 on page 279 Version 5 0 175 December 2006 r wi AudioCodes MediaPack 5 6 5 7 Configuring the IKE Table Use the IKE Table screen to configure the IKE parameters For detailed information on IPSec and IKE refer to Section 12 1 on page 279 5 6 6 Configuring the Management Settings gt To configure the Management Settings parameters take these 4 steps 1 Open the Management Settings screen Advanced Configuration menu gt Management Settings the Management Settings screen is displayed Figure 5 51 Management Settings Screen Management Settings Syslog Settings syslog Server IP Address 10 6 2 19 Syslog Server Port Enable Syslog SNMP Managers Table SNMP Community String SNMP V3 Table Enable SNMP Trap Manager Host Name Activity Types to Report via Activity Log Messages Parameters Value Change Auxiliary Files Loading Device Reset Flash Memory Burn
102. of NATs and firewalls located between it and the public Internet It provides the gateway 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 gateway It also discovers the binding lifetime of the NAT the refresh rate necessary to keep NAT Pinholes open On startup the gateway 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 NATBindingDefaultTimeout At the beginning of each call if STUN is needed 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 To enable STUN m Set the parameter EnableSTUN to 1 E Define the STUN server s address by performing one of the following e Define the STUN server s address by assigning it a domain name The STUN client can perform the required SRV query to resolve it to an IP address and port sort the server list and use the servers according to the sorted list e Determine the IP address of the primary and optionally the secondary STUN servers using the parameters STUNServerPrimarylP and STUNServerSecondarylP If the primary STUN server isn t available the gateway tries to communicate
103. of the TFTP utility that is used for file transfer of software and initialization files to the gateway When creating a new client this field is populated with the IP address used by the BootP Tool If a different TFTP utility is to be used change the IP address in this field to the IP address used by the other utility Version 5 0 357 December 2006 Ta WH wt AudioCodes MediaPack Boot File This field specifies the file name for the software cmp file that is loaded by the TFTP utility to the VoIP gateway after the VoIP gateway receives the BootReply message The actual software file is located in the TFTP utility directory that is specified in the BootP Preferences window The software file can be followed by command line switches For information on available command line switches refer to Section C 11 6 on page 359 Once the software file loads into the gateway the gateway begins functioning from that software In order to save this software to non volatile memory only the cmp file i e the compressed firmware file can be burned to your device s flash memory the fb flag must be added to the end of the file name If the file is not saved the gateway reverts to the old version of software after the next reset The Boot file field can contain up to two file names cmp file name to be used for load of application image and ini file name to be used for gateway provisioning Either one two or no file names can a
104. older parameter not the table parameter two more parameters are SET to ENABLE snmpManagerlsUsed 0 and snmpManagerTrapSendingEnable 0 are both set to 1 SNMP NAT Traversal A NAT placed between the gateway and the element manager calls for traversal solutions Trap source port all traps are sent out from the SNMP port default 161 A manager receiving these traps can use the binding information in the UDP layer to traverse the NAT back to the device The trap destination address port and IP are as configured in the snmpTargetMIB acKeepAliveTrap this trap is designed to be a constant life signal from the device to the manager allowing the manager NAT traversal at all times The acBoardTrapGlobalsAdditionallnfo1 varbind has the device s serial number The destination port i e the manager port for this trap can be set to be different than the port to which all other traps are sent To do this use the acSysSNMPkKeepAliveTrapPort object in the acSystem MIB or the ini file parameter KeepAliveTrapPort The trap is instigated in three ways e Via an ini file parameter SendKeepAliveTrap 1 This ensures that the trap is continuously sent The frequency is set via the 9 10 of the acSysSTUNBindingLifeTime object e After the STUN client has discovered a NAT any NAT If the STUN client cannot contact a STUN server Note The two latter options require the STUN client be enabled EnableSTUN Also once the acKee
105. or any other auxiliary file listed in the Wizard is optional Figure 5 67 Load a CPT File Screen CMP file Load a CPT file from your computer to the device INI file Browse _ J Use existing file usa_tones_11 dat Call Progress Tones File SIP User s Manual 200 Document LTRT 65408 SIP User s Manual 5 Web Management 9 Follow the same procedure you followed when loading the ini file refer to Step 7 The same procedure applies to the Load a coefficient file screen 10 In the Finish screen refer to Figure 5 68 the Next button is disabled Complete the upgrade process by clicking Reset or Cancel e Click Reset the MediaPack burns the newly loaded files to flash memory The Burning files to flash memory screen appears Wait for the burn to finish When it finishes the End Process screen appears displaying the burned configuration files refer to Figure 5 69 e Click Cancel the MediaPack resets utilizing the files previously stored in flash memory Note that these are NOT the files you loaded in the previous Wizard steps Figure 5 68 Finish Screen You have finished the upgrade process Now click the Reset button to burn the configuration to the device flash memory and restart the device or click the Cancel button to restart the device with the previously flash burned configuration Figure 5 69 End Process Screen CMP Version ID 5 004 001 003 C
106. parameter containing its URL m To load different configurations ini files for specific gateways add the string lt MAC gt to the URL This mnemonic is replaced with the MediaPack hardware MAC address Resulting in an ini file name request that contains the gateway s MAC address To automatically update the cmp file use the parameter CmpFileURL to specify its name and location As a precaution to protect the MediaPack from an accidental update the Automatic Update mechanism doesn t apply to the cmp file by default Therefore to enable it set the parameter AutoUpdateCmpFile to 1 Version 5 0 263 December 2006 Ta WH wt AudioCodes MediaPack The following example illustrates how to utilize Automatic Updates for deploying devices with minimum manual configuration gt To utilize Automatic Updates for deploying the MediaPack with minimum manual configuration take these 5 steps 1 Set up a Web server in the following example it is http www corp com where all configuration files are to be stored 2 To each device pre configure the following parameter DHCP DNS are assumed IniFile URL http www corp com master_configuration ini 3 Create a file named master_configuration ini with the following text Common configuration for all devices CpetFileURL http www corp com call_progress dat Check for updates every 60 minutes AutoUpdateFrequency 60 Additional configurat
107. parameter in the Via header 0 Enabled 1 Disabled default The gateway adds an rport parameter to the Via header field of each outgoing SIP message The first Proxy that receives this message sets the rport value of the response to the actual port from which the request was received This method is used for example to enable the gateway to identify its port mapping outside a NAT If the Via doesn t include rport tag the destination port of the response will be taken from the host part of the VIA If the Via includes rport tag with no port value the destination port of the response will be the source port of the incoming request If the Via includes rport tag with a port value rport 1001 the destination port of the response will be the port indicated in the rport tag Serial parameters applicable only to the VM SMDI application SerialBaudRate SerialData SerialParity SerialStop SerialFlowControl Determines the value of the RS 232 baud rate The valid range is any value It is recommended to use the following standard values 1200 2400 9600 default 14400 19200 38400 57600 115200 Determines the value of the RS 232 data bit 7 7 bit 8 8 bit default Determines the value of the RS 232 polarity 0 None default 1 Odd 2 Even Determines the value of the RS 232 stop bit 1 1 bit default 2 2 bit Determines the value of the RS
108. performed on Proxy servers that are marked as Online All outgoing messages are equally distributed across the Proxy IP list REGISTER messages are also distributed unless a RegistrarIP is configured The Proxy IP list is refreshed according to ProxylPListRefreshTime If a change in the order of the entries in the list occurs all load statistics are erased and balancing starts over again When Random Weights 2 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 gateway sends the requests in such a fashion that each Proxy receives a percentage of the requests according to its assigned weight Load Balancing is not used in the following scenarios The ProxylP table includes more than one entry The only Proxy defined is an IP address and not an FQDN SRV usage is not enabled DNSQueryType The SRV response includes several records with a different Priority value Defines the time interval in seconds between refreshes of the Proxy IP list This parameter is used only when ProxyLoadBalancingMethod 1 The interval range is 5 to 2 000 000 The default interval is 60 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 Contact and Record Route headers Valid options include 0 A Record default
109. refer to Section 5 10 3 on page 206 When positioning your curser over a parameter name or a table for more than 1 second a short description of this parameter is displayed Note that parameters preceded by an exclamation mark are not changeable on the fly and require that the device be reset 5 4 2 Saving Changes To save changes to the volatile memory RAM click the Submit button changes to parameters with on the fly capabilities are immediately available other parameter are updated only after a gateway reset Parameters that are only saved to the volatile memory revert to their previous settings after hardware reset When performing a software reset i e via Web or SNMP you can choose to save the changes to the non volatile memory To save changes so they are available after a power fail you must save the changes to the non volatile memory flash To save the changes to flash refer to Section 5 10 2 on page 205 Version 5 0 53 December 2006 r wi AudioCodes MediaPack 5 4 3 5 4 4 Entering Phone Numbers in Various Tables Phone numbers entered into various tables on the gateway such as the Tel to IP routing table must be entered without any formatting 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 does not work The hyphen character is used in number entry only as part of a range definitio
110. refer to Section 5 3 on page 51 2 Inthe URL field append the suffix AdminPage note that it s case sensitive to the IP address e g http 10 1 229 17 AdminPage 3 Click the INI Parameters option the INI Parameters screen is displayed shown in a Figure 10 8 Figure 10 8 INI Parameters Screen Parameter name Enter value Parameter name Enter value 3GDelErrarSDU Apply new value OUTPUT WINDOW 4 From the Parameter Name drop down list select the required ini file parameter 5 Inthe Enter Value field enter the parameter s new value 6 Click the Apply New Value button the INI Parameters screen is refreshed the parameter name with the new value appears in the fields at the top of the screen and the Output Window pane displays a log displaying information on the operation Note You cannot load the image files e g logo background image files to the device by choosing a file name parameter in this screen SIP User s Manual 276 Document LTRT 65408 SIP User s Manual 11 Special Applications Metering Tones Relay 11 Special Applications Metering Tones Relay The MediaPack FXS and FXO gateways can be used to relay standard 12 or 16 kHz metering tones over the IP network as illustrated in Figure 11 1 below Figure 11 1 Metering Tone Relay Architecture After a call is established between the FXS and FXO gateways the PSTN generates 12 or 16 kHz metering tones tow
111. required to activate a loaded CPT file and may be required for the activation of certain ini file parameters To save the loaded auxiliary files so they are available after a power fail refer to Section 5 10 2 on page 205 To reset the MediaPack refer to Section 5 10 3 on page 206 SIP User s Manual 202 Document LTRT 65408 SIP User s Manual 5 Web Management Figure 5 70 Auxiliary Files Screen Auxiliary Files Send FXS Coefficient file from your computer to the device Browse _ Send File Send Voice Prompts file from your computer to the device Browse Send File Send Call Progress Tones file from your computer to the device Browse _ Send File Send Prerecorded Tones file from your computer to the device Browse _ Send File Send User Info file from your computer to the device Browse _ Send File 5 9 2 1 Loading the Auxiliary Files via the ini File gt To load the auxiliary files via the ini file take these 3 steps 1 In the ini file define the auxiliary files to be loaded to the MediaPack You can also define in the ini file whether the loaded files should be stored in the non volatile memory so that the TFTP process is not required every time the MediaPack boots up 2 Locate the auxiliary files you want to load and the ini file in the same directory 3 Invoke a BootP TFTP session the ini and auxiliary files are loaded o
112. s Manual 148 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 1 6 Network Settings ini File Parameters Table 5 41 describes the Network parameters that can only be configured via the ini file Table 5 41 Network Settings ini File Parameters continues on pages 149 to 151 ini File Parameter Name EnablePPPoE PPPoEUserName PPPoEPassword PPPoEServerName PPPoEStaticlPAddress PPPoERecoverlPAddress PPPoERecoverSubnetMask PPPoERecoverDfgwAddress PPPoELCPEchoEnable EthernetPhyConfiguration Valid Range and Description Enables the PPPoE Point to Point Protocol over Ethernet feature 0 Disable default 1 Enable User Name for PAP or Host Name for CHAP authentication The valid range is a string of up to 47 characters The default value is 0 Password for PAP or Secret for CHAP authentication The valid range is a string of up to 47 characters The default value is 0 Server Name for CHAP authentication The valid range is a string of up to 47 characters The default value is 0 IP address to use in a static configuration setup If set used during PPP negotiation to request this specific IP address from the PPP server If approved by the server this IP address is used during the session The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 0 0 0 0 IP address to use when booting from the flash to non PPPoE Point to Point Protocol over Ethernet
113. scroll down until the HTTP 1 1 Settings are displayed and verify that the Use HTTP 1 1 option is checked 3 Restart the browser 5 4 Getting Acquainted with the Web Interface Figure 5 2 shows the general layout of the Web Interface screen Figure 5 2 MediaPack Web Interface e g MP 118 FXS Fa w k w AudioCodes MP 118 FXS eoo oee e a l Quick Setup Advanced Configuration Status amp Diagnostics Software Update Maintenance Log Off SIP User s Manual 52 Document LTRT 65408 SIP User s Manual 5 Web Management The Web Interface screen features the following components H Title bar contains three configurable elements corporate logo a background image and the product s name For information on how to modify these elements refer to Section 10 5 on page 267 m Product name the gateway model name m Main menu bar always appears on the left of every screen to quickly access parameters submenus submenu options functions and operations E Submenu bar appears on the top of screens and contains submenu options Main action frame the main area of the screen in which information is viewed and configured Home icon when clicked it opens the Trunk amp Channel Status screen refer to Section 5 8 on page 195 Corporate logo AudioCodes corporate logo For information on how to remove this logo refer to Section 10 5 on page 267 Search Engine for searching ini file par
114. sent and HookFlashOption received Valid options include Not Supported 0 Hook Flash indication isn t sent default INFO 1 Send proprietary INFO message with Hook Flash indication RFC 2833 4 RFC 2833 Note FXO gateways support the receiving of RFC 2833 Hook Flash signals IsHookFlashUsed This parameter is obsolete use instead the parameter HookFlashOption Version 5 0 T9 December 2006 Ta WH wt AudioCodes MediaPack Table 5 7 DTMF amp Dialing Parameters continues on pages 74 to 76 Parameter Description Digit Mapping Rules Digit map pattern If the digit string dialed number has matched one of the DigitMapping patterns in the digit map the gateway stops collecting digits and starts to establish a call with the collected number The digit map pattern contains up to 52 options separated by a vertical bar and enclosed in parenthesis The maximum length of the entire digit pattern is limited to 152 characters Available notations n m represents a range of numbers single dot represents repetition x represents any single digit T represents a dial timer configured by TimeBetweenDigits parameter S should be used when a specific rule that is part of a general rule is to be applied immediately For example if you enter the general rule x T and the specific rule 11x you should append S to the specific rule 11xS For example 11xS OOT 1 7 xxx 8xxxxxXxx XXX
115. set e TxDTMFOption 4 1 to 5 DTMF Option RFC 2833 e RxDTMFOption 3 Declare RFC 2833 in SDP Yes Note that to set the RFC 2833 payload type with a different value other than its default 96 configure the RFC2833PayloadType RFC 2833 Payload Type parameter The gateway negotiates the RFC 2833 payload type using local and remote SDP and sends packets using the PT from the received SDP The gateway expects to receive RFC 2833 packets with the same PT as configured by the RFC2833PayloadType parameter If the remote side doesn t include telephony event in its SDP the gateway sends DTMF digits in transparent mode as part of the voice stream 5 Sending DTMF digits in RTP packets as part of the audio stream DTMF Relay is disabled Note that this method is normally used with G 711 coders with other LBR coders the quality of the DTMF digits is reduced To ser this mode e TxDTMFOption 0 1 to 5 DTMF Option Disable e RxDTMFOption 0 Declare RFC 2833 in SDP No e DTMFTransportType 2 DTMF Transport Type Transparent DTMF The gateway is always ready to receive DTMF packets over IP in all possible 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 gateway s SDP set RxDTMFOption 0 in the ini file The following parameters affect
116. the IPSec mechanism is IPSecPolicyProtocol applied to O Any protocol default 17 UDP 6 TCP Or any other protocol type defined by IANA Internet Assigned Numbers Authority Related Key Exchange Method Determines the index for the corresponding IKE entry Note that several Inger policies can be associated with a single IKE entry IPsecPolicyKeyExchangeMet The valid range is O to 19 The default value is 0 hodindex IKE Second Phase Parameters Quick Mode SA Lifetime sec Determines the time in seconds the SA negotiated in the second IKE session I PsecPolicyLifelnSec quick mode is valid After the time expires the SA is re negotiated The default value is 28800 8 hours SA Lifetime KB Determines the lifetime in kilobytes the SA negotiated in the second IKE IPSecPolicyLifelnKB session quick mode is valid After this size is reached the SA is re negotiated The default value is 0 this parameter is ignored The lifetime parameters IPsecPolicyLifelnSec and IPSecPolicyLifelnKB determine the duration an SA is valid When the lifetime of the SA expires it is automatically renewed by performing the IKE second phase negotiations To refrain from a situation where the SA expires a new SA is being negotiated while the old one is still valid As soon as the new SA is created it replaces the old one This procedure occurs whenever an SA is about to expire SIP User s Manual 284 Document LTRT 65408 SIP Use
117. the RJ 11 sockets on the front of an MDF Adaptor Block as shown in Figure 3 13 on page 39 RS 232 serial For detailed information on connecting the MP 124 RS 232 port to your PC refer to Section cable 3 2 4 1 on page 40 AC Power cable Connect the MP 124 power socket to the mains MP 124 Safety Notice To protect against electrical shock and fire use a 26 AWG min wire to connect analog FXS lines to the 50 pin Telco connector Figure 3 11 RJ 45 Ethernet Connector Pinouts RJ 45 Connector and Pinout 12345678 1 Txt ate 578 3 Rxt 6 Rx connected Figure 3 12 50 pin Telco Connector MP 124 FXS only Pin Numbers SIP User s Manual 38 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack Figure 3 13 MP 124 in a 19 inch Rack with MDF Adaptor FRONT INPUT 19 inch Rack 24 line cords 2 wire with RJ 11 Rear View connectors M D F Adaptor Block rear ee ee ee ee a m Pe A ae REAR OUTPUT 24 wire pairs in LAN Cable Octopus cable with 50 pin male Telco connector AC Power Cord Connect to AA Grounding Strap 50 pin female Telco connector RS 232 Cable Table 3 5 Pin Allocation in the 50 pin Telco Connector Phone Channel Connector Pins Phone Channel Connector Pins 1 1 26 13 13 38 2 2 27 14 14 39 3 3 28 15 15 40 4 4 29 16 16 41 5 5 30 17 17 42 6 6 31 18 18 43 7 7 32 19 19 44 8 8 33 20 20 45 9 9 34 21 21 46 10 10 35 22 22 47 11 11 36 23 23 48 12 12 37 24 24
118. the gateway After the gateway is powered up it attempts to communicate with a BootP server If a BootP server is not responding and if DHCP is enabled then the gateway attempts to get its IP address and other network parameters from the DHCP server Note After you enable the DHCP Server from the Web browser follow this procedure a Click the Submit button Save the configuration using the Maintenance button before you reset the gateway For information on how to save the configuration refer to Section 5 10 2 on page 205 Reset the gateway directly Web reset doesn t trigger the BootP DHCP procedure and the parameter DHCPEnable reverts to 0 Note that throughout the DHCP procedure the BootP TFTP application must be deactivated Otherwise the MediaPack receives a response from the BootP server instead of the DHCP server Note For additional information on DHCP refer to Section 7 2 on page 212 ini file note The DHCPEnable 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 Global gateway IP address Define if static Network Address Translation NAT device is used between the gateway and the Internet For detailed information on IP QoS via Differentiated Services refer to Section 9 8 on page 253 Network QoS NetworkServiceClassDiffSer v Media Premium QoS PremiumServiceClassMedi
119. the gateway 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 gateway on the same port number the gateway accepts only one of the RTP streams and rejects the rest of the streams The RTP stream is selected according to the following procedure 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 accepted stream If a new packet arrives whose source IP address or UDP port are different to the currently accepted RTP stream there are two options E The new packet has a source IP address and UDP port which are the same as the remote IP address and UDP port that were stated during the opening of the channel In this case the gateway reverts to this new RTP stream m The new packet has any other source IP address and UDP port in which case the packet is dropped Multiple Routers Support Multiple routers support is designed to assist the media gateway when it operates in a multiple routers network The gateway learns the network topology by responding to ICMP redirections and caches them as routing rules with expiration time When a set of routers operating within the same subnet serve as gateways to that network and intercommunicate using a dynamic routing protocol the routers can determine the
120. the way the MediaPack SIP handles the DTMF digits TxDTMFOption RXxXDTMFOption and RFC2833PayloadType described in Table 5 7 MGCPDTMFDetectionPoint DTMFDigitLength and DTMFinterDigitInterval Table 5 47 E DTMFVolume and DTMFTransportType Table 5 42 Version 5 0 219 December 2006 Ta WH wt AudioCodes MediaPack 8 3 8 3 2 Fax amp Modem Transport Modes Fax Modem Settings Users may choose to use one of the following transport methods for fax and for each modem type V 22 V 23 Bell V 32 V 34 E Fax relay demodulation modulation E Bypass using a high bit rate coder to pass the signal E Transparent passing the signal in the current voice coder When the fax relay mode is enabled distinction between fax and modem is not immediately possible at the beginning of a session The channel is therefore in Answer Tone mode until a distinction is determined The packets being sent to the network at this stage are T 38 compliant fax relay packets Configuring Fax Relay Mode When FaxTransportMode 1 relay mode then on detection of fax the channel automatically switches from the current voice coder to answer tone mode and then to T 38 compliant fax relay mode When fax transmission has ended the reverse switching from fax relay to voice is performed This mode switching automatically occurs at both the local and remote endpoints Users can limit the fax rate using the FaxRelayMaxRate parameter a
121. this manual Hexadecimal notation is indicated by Ox preceding the number Related Documentation Document Manual Name LTRT 656xx e g LTRT 65601 MP 11x amp MP 124 SIP Release Notes LTRT 598xx MP 11x amp MP 124 MGCP H 323 SIP Fast Track Guide LTRT 665xx CPE Configuration Guide for IP Voice Mail Version 5 0 7 December 2006 aN bil gt udioCodes MediaPack Note 1 MediaPack refers to the MP 124 MP 118 MP 114 and MP 112 VoIP gateways Note 2 MP 11x refers to the MP 118 MP 114 and MP 112 VoIP gateways Where network appears in this manual it means Local Area Network LAN Wide Area Network WAN etc accessed via the gateway s Ethernet interface 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 VoIP gateway interfaces between the CO PBxX 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 gateway interfaces between the analog telephone devices and the Internet W
122. thoroughly when EnableDiagnostics is set to 2 flash is only partially tested While the Detailed test is running the Ready and Fail LEDs are lit If an error is detected an error message is sent to the Syslog Warning To continue regular operation the Detailed test must be disabled Set the parameter EnableDiagnostics to 0 and reset the MediaPack MediaPack Line Testing The MediaPack features a mechanism that performs tests on the telephone lines connected to FXS and FXO ports These tests provide various line measurements The line testing mechanism is performed on channel 1 and therefore disconnects any call that is in progress on this channel Therefore it s recommended to perform the testing only when there are no calls in progress The following line tests are available on FXS gateways m Hardware revision number Temperature above or below limit only if a thermometer is installed m Hook state Version 5 0 301 December 2006 Ta A wi AudioCodes MediaPack Coefficients checksum Message waiting indication status Ring state Reversal polarity state Line current only on port 0 Line voltage between TIP and RING only on port 0 3 3 V reading only on port 0 Ring voltage only on port 0 Long line current only on port 0 The following line tests are available on FXO gateways Hardware revision number Hook state m Reversal polarity state gt To perform analog
123. to No Answer Number of Failed Calls due to No Route Number of Failed Calls due to No Matched Capabilities Number of Failed Calls due to Other Failures n Average Call Duration sec Attempted Fax Calls Counter oo Oo 0 0 gt Successful Fax Calls Counter Table 5 59 Call Counters Description continues on pages 189 to 190 Counter Description This counter indicates the number 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 five failed call counters Only one of the established failed call counters is incremented every time This counter 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 bigger then 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
124. to the second Proxy Note 1 If EnableProxyKeepAlive 1 or 2 the gateway monitors the connection with the Proxies by using keep alive messages OPTIONS or REGISTER Note 2 To use Proxy Redundancy you must specify one or more redundant Proxies using multiple ProxylIP lt IP address gt definitions Note 3 When port number is specified e g domain com 5080 DNS SRV queries aren t performed even if ProxyDNSQueryType is set to 1 Use this parameter to assign a name to the gateway e g gateway1 com Ensure that the name you choose is the one that the Proxy is configured with to identify your media gateway Note If specified the gateway Name is used as the host part of the SIP URI in the From header If not specified the gateway IP address is used instead default Defines the user name that is used in From and To headers of REGISTER messages Applicable only to single registration per gateway AuthenticationMode 1 If GWRegistrationName isn t specified default the Username parameter is used instead Note If AuthenticationMode 0 all the gateway s endpoints are registered with a user name that equals to the endpoint s phone number SIP User s Manual 66 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 4 Proxy amp Registration Parameters continues on pages 66 to 71 Parameter First Redundant Proxy IP Address ProxyIP Second Redund
125. to which this trap relates For devices where there are no chassis options the slot number of the gateway is always 1 Alarm Traps The following tables provide information on alarms that are raised as a result of a generated SNMP trap The component name described in each of the following headings refers to the string that is provided in the acBoardTrapGlobalsSource trap varbind To clear a generated alarm the same notification type is sent but with the severity set to cleared F 1 1 Component Board lt n gt The source varbind text for all the alarms under this component is Board lt n gt where n is the slot number For MediaPack lt n gt 1 Table F 1 acBoardFatalError Alarm Trap Alarm acBoardFatalError OID 1 3 6 1 4 1 5003 9 10 1 21 2 0 1 Default Severity Critical Event Type equipmentAlarm Probable Cause underlyingResourceUnavailable 56 Alarm Text Board Fatal Error lt text gt Status Changes Condition Any fatal error Alarm status Critical lt text gt value A run time specific string describing the fatal error Condition After fatal error Alarm status Corrective Action Status stays critical until reboot A clear trap is not sent Capture the alarm information and the Syslog clause if active Contact your first level support group The support group will likely want to collect additional data from the device and perform a reset Version 5 0 375 December 2006 Ta fal AudioCodes
126. typical MediaPack VoIP application Figure 1 1 Typical MediaPack VoIP Application F Router F 4E naw pl 1 2 SIP Overview SIP Session Initialization Protocol is an application layer control signaling protocol used on the MediaPack 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 on the MediaPack complies with the IETF Internet Engineering Task Force RFC 3261 refer to http www ietf org SIP User s Manual 20 Document LTRT 65408 SIP User s Manual 1 Overview 1 3 MediaPack Features This section provides a high level overview of some of the many MediaPack supported features 1 3 1 General Features Superior high quality Voice Data and fax over IP networks Toll quality voice compression Enhanced capabilities including MWI long haul metering CID and out door protection Proven integrat
127. useful when the gateway does not yet have an IP address and therefore cannot respond to an ARP Because this feature creates an entry in the computer ARP cache Administrator Privileges are required If the computer is not set to allow administrator privileges ARP Manipulation cannot be enabled ARP Manipulation Enabled Enable ARP Manipulation to remotely reset a gateway that does not yet have a valid IP address If ARP Manipulation is enabled the following two commands are available e Reply Type Reply to a BootRequest can be either Broadcast or Unicast The default for the BootP Tool is Broadcast In order for the reply to be set to Unicast ARP Manipulation must first be enabled This then enables the BootP Tool to find the MAC address for the client in the ARP cache so that it can send a message directly to the requesting device Normally this setting can be left at Broadcast e ARP Type The type of entry made into the ARP cache on the computer once ARP Manipulation is enabled can be either Dynamic or Static Dynamic entries expire after a period of time keeping the cache clean so that stale entries do not consume computer resources The Dynamic setting is the default setting and the setting most often used Static entries do not expire Number of Timed Replies This feature is useful for communicating to VoIP gateways that are located behind a firewall that would block their BootRequest messages from getting through to the com
128. v Protocol version Yes o Owner creator and session identifier Yes a Attribute information Yes c Connection information Yes d Digit Yes m Media name and transport address Yes s Session information Yes t Time alive header Yes b Bandwidth header Yes u Uri Description Header Yes e Email Address header Yes i Session Info Header Yes p Phone number header Yes y Year Yes Version 5 0 343 December 2006 Ta WH wt AudioCodes MediaPack B 5 SIP Responses The following SIP responses are supported by the gateway 1xx Response Information Responses 2xx Response Successful Responses 3xx Response Redirection Responses 4xx Response Client Failure Responses 5xx Response Server Failure Responses 6xx Response Global Responses B 5 1 1xx Response Information Responses Table B 5 1xx SIP Responses 1xx Response Supported Comments 100 Trying Yes The SIP gateway generates this response upon receiving of Proceeding message from ISDN or immediately after placing a call for CAS signaling 180 Ringing Yes The SIP gateway generates this response for an incoming INVITE message On receiving this response the gateway waits for a 200 OK response 181 Call is Yes The SIP gateway does not generate these responses However the being gateway does receive them The gateway processes these responses forwarded the same way that it processes the 100 Trying response 182 Queued Yes The SIP gateway gene
129. when you are finished editing templates To delete an existing template take these 3 steps Select the template by clicking its name from the list of templates in the window Click the Delete Current Template button E a warning pop up message appears To delete the template click Yes Note that if this template is currently in use the template cannot be deleted SIP User s Manual 360 Document LTRT 65408 SIP User s Manual D RTP RTCP Payload Types and Port Allocation D RTP RTCP Payload Types and Port Allocation RTP Payload Types are defined in RFC 3550 and RFC 3551 We have added new payload types to enable advanced use of other coder types These types are reportedly not used by other applications D 1 Packet Types Defined in RFC 3551 Table D 1 Packet Types Defined in RFC 3551 Payload Type Description Basic Packet Rate msec 0 G 711 Law 10 20 2 G 726 32 10 20 4 G 723 6 3 5 3 kbps 30 8 G 711 A Law 10 20 18 G 729A B 20 200 RTCP Sender Report Randomly approximately every 5 seconds when packets are sent by channel 201 RTCP Receiver Report Randomly approximately every 5 seconds when channel is only receiving 202 RTCP SDES packet 203 RTCP BYE packet 204 RTCP APP packet D 2 Defined Payload Types Table D 2 Defined Payload Types Payload Type Description Basic Packet Rate msec 96 RFC 2833 DTMF relay 20 102 Fax Bypass 20 103 Modem Bypass 20 104 RFC 2198 Redundancy Same as channel s voice co
130. with a Proxy server set Working with Proxy field to Yes and enter the IP address of the primary Proxy server in the field Proxy IP Address When no Proxy is used the internal routing table is used to route the calls Enter the Proxy Name in the field Proxy Name If Proxy name is used it 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 SIP User s Manual 46 Document LTRT 65408 SIP User s Manual 4 Getting Started 9 Configure Enable Registration to Yes or No e No the MediaPack does not register to a Proxy server Registrar default e Yes the MediaPack registers to a Proxy server Registrar at power up and every Registration Time seconds The MediaPack sends a REGISTER request according to the Authentication Mode parameter For detailed information on the parameters Registration Time and Authentication Mode refer to Table 5 4 on page 66 To configure the Coders Table click the arrow button next to Coders Table For information on how to configure the Coders Table refer to Section 5 5 1 3 on page 72 To configure the Tel to IP Routing Table click the arrow button next to Tel to IP Routing Table For information on how to configure the Tel to IP Routing Table refer to Section 5 5 5 2 on page 100 To configure the
131. 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 t 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 Table 5 15 Phone Context Parameters Parameter Description Add Phone Context As Prefix Determines whether or not the received Phone Context parameter is added as a AddPhoneContextAsPrefix prefix to the outgoing ISDN SETUP Called and Calling numbers Valid options include 0 Disable default 1 Enable NPI Select the Number Plan assigned to this entry You can select the following 0 Unknown default 1 E 164 Public 9 Private TON Select the Number Type assigned to this entry f you selected Unknown as the NPI you can select Unknown 0 If you selected Private as the NPI you can select Unknown 0 Level 2 Regional 1 Level 1 Regional 2 PSTN Specific 3 or Level 0 Regional Local 4 If you selected E 164 Public as the NPI you can select Unknown 0 International 1 National 2 Network Specific 3 Subscriber 4 or Abbreviated 6 Phone Context The Phone Context SIP URI parameter Version 5 0 97 December 2006 Ta A wi AudioCodes MediaPack 5 5 5 Configuring the Routing Tables Use this submenu to configure the gateway s IP
132. with the secondary server m Use the parameter NATBindingDefaultTimeout to determine the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires STUN only applies to UDP doesn t support TCP and TLS STUN can t be used when the gateway is located behind a symmetric NAT For defining the STUN server s address either use the stunServerPrimarylpAddress or the StunServerDomainName method with priority to the first one SIP User s Manual 248 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities 9 2 2 9 2 3 9 3 First Incoming Packet Mechanism If the remote gateway resides behind a NAT device it s possible that the MediaPack can activate the RTP RTCP T 38 streams to an invalid IP address UDP port To avoid such cases the MediaPack automatically compares the source address of the incoming RTP RTCP T 38 stream with the IP address and UDP port of the remote gateway 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 addresses and UDP ports Users can choose to disable the NAT mechanism by setting the ini file parameter DisableNAT to 1 The two parameters EnablelpAddrTranslation and EnableUdpPortTranslation enable users to specify the type of compare operation that takes place on the
133. you must have administrator privileges for the computer you are using SIP User s Manual 356 Document LTRT 65408 SIP User s Manual C BootP TFTP Configuration Utility C 11 4 Testing the Client There should only be one BootP utility supporting any particular client MAC active on the network at any time gt To check if other BootP utilities support this client take these 4 steps Select the client that you wish to test by clicking the client name in the main area of the Client Configuration Window Click the Test Selected Client button KEF Examine the Log Window on the Main Application Screen If there is another BootP utility that supports this client MAC there is a response indicated from that utility showing the status Listed At along with the IP address of that utility If there is another utility responding to this client you must remove that client from either this utility or the other one C 11 5 Setting Client Parameters Client parameters are listed on the right side of the Client Configuration Window Client MAC The Client MAC is used by BootP to identify the VoIP gateway The MAC address for the VoIP gateway is printed on a label located on the VoIP gateway hardware Enter the Ethernet MAC address for the VoIP gateway in this field Click the box to the right of this field to enable this particular client in the BootP tool if the client is disabled no replies are sent to BootP requests Note When
134. 0 default gateway IP address is 0 0 0 0 Configuration Concepts Users can utilize the MediaPack in a wide variety of applications enabled by its parameters and configuration files e g Call Progress Tones CPT The parameters can be configured and configuration files can be loaded using E A standard Web Browser described and explained in Chapter 5 on page 49 A configuration file referred to as the ini file For information on how to use the ini file refer to Chapter 6 on page 209 E An SNMP browser software refer to Chapter 14 on page 305 m AudioCodes Element Management System EMS refer to Section 14 10 on page 322 and to AudioCodes EMS User s Manual or EMS Product Description To upgrade the MediaPack load new software or configuration files onto the gateway use the Software Upgrade wizard available through the Web Interface refer to Section 5 9 1 on page 197 or alternatively use the BootP TFTP configuration utility refer to Section 7 3 1 on page 213 For information on the configuration files refer to Chapter 6 on page 209 Assigning the MediaPack IP Address To assign an IP address to the MediaPack use one of the following methods m HTTP using a Web browser refer to Section 4 2 1 below E BootP refer to Section 4 2 2 on page 42 m Voice Menu using a standard touch tone telephone connected to one of the FXS analog ports refer to Section 4 2 3 on page 43 This method doesn t apply to FXO
135. 0 0 0 0 IP Address f 0 32 174 50 Subnet Mask 255 255 0 0 Default Gateway Address Media Network Settings 0 0 0 0 IP Address f 0 33 174 50 Subnet Mask 255 255 0 0 Default Gateway Address j10 33 0 1 Click the Submit button to save your changes 5 Configure the IP Routing table by completing the following steps the IP Routing table is required to define static routing rules for the OAM and Control networks since a default gateway isn t supported for these networks e Open the IP Routing Table screen Advanced Configuration menu gt Network Settings gt IP Routing Table option the IP Routing Table screen is displayed Figure 9 4 Example of the IP Routing Table Screen Routing Table Delete Row _ Destination IP Address Destination Mask Gateway IP Address Hop Count Interface 103200 pesoss00 froszivaso pravaessay fp con 7000 pooo fzor pimes h jom a e Use the Add a new table entry pane to add the routing rules shown in Table 9 3 below Table 9 3 Example of IP Routing Table Configuration Destination IP TE Gateway IP Network Address Destination Mask Address Hop Count Type 130 33 4 6 255 255 255 255 10 32 0 1 20 Control 83 4 87 6 255 255 255 0 10 31 0 1 20 OAM e Click the Submit button to save your changes 6 Save your changes to flash so they are available after a power fail refer to Section 5 10 2 on page 205 7 Reset the gate
136. 024 3DES MD5 1024 3DES SHA1 786 3DES MD5 786 gt To configure the IKE table using the ini file The IKE parameters are configured using ini file tables described in Section 10 5 on page 267 Each line in the table refers to a different IKE peer The Format line IKE_DB_INDEX in the example below specifies the order in which the actual data lines are written The order of the parameters is irrelevant Parameters are not mandatory unless stated otherwise To support more than one Encryption Authentication DH Group proposals for each proposal specify the relevant parameters in the Format line Note that the proposal list must be contiguous SIP User s Manual 282 Document LTRT 65408 SIP User s Manual 12 Security Figure 12 2 Example of an IKE Table IPSec_IKEDB_Table Format IKE_DB_INDEX IKEPolicySharedKey IKEPolicyProposalEncryption_0O IKEPolicypRoposalAuthentication_0 IKEPolicyProposalDHGroup_0 IKEPolicyProposalEncryption_1l IKEPolicypRoposalAuthentication_l IKEPolicyProposalDHGroup_1l IKEPolicyLifeInSec IkePolicyAuthenticationMethod IPSEC_IKEDB_TABLE 0 123456789 1 2 i 266007 Op IPSEC_IKEDB_TABLE In the example above a single IKE peer is configured and a Pre shared key authentication is selected Its pre shared key is 123456789 Two security proposals are configured DES SHA1 786DH and 3DES SHA1 1024DH gt To configure the IKE table using the Embedded Web Server take these 6
137. 08 Bad Request Unauthorized Payment Required Forbidden Not Found Method Not Allowed Not Acceptable Proxy Authentication Required Request Timeout Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Comments The gateway does not generate this response On reception of this message before a 200 OK has been received the gateway responds with an ACK and disconnects the call Authentication support for Basic and Digest On receiving this message the GW issues a new request according to the scheme received on this response The gateway does not generate this response On reception of this message before a 200 OK has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 200 OK has been received the gateway responds with an ACK and disconnects the call The SIP gateway generates this response if it is unable to locate the callee On receiving this response the gateway notifies the User with a Reorder Tone The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call Authentication support for Basic and Digest On
138. 1 Reason 2 Reason 3 Reason 4 2 In the IP to Tel Reasons table from the drop down list select up to 4 different call failure reasons that invoke an alternative IP to Tel routing 3 In the Tel to IP Reasons table from the drop down list select up to 4 different call failure reasons that invoke an alternative Tel to IP routing Version 5 0 107 December 2006 Ta WH wt AudioCodes MediaPack 4 Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 21 Reasons for Alternative Routing ini File Parameter Parameter Name in ini File Parameter Format AltRouteCauseTel2IP AltRouteCauseTel2IP lt SIP Call failure reason from IP gt For example AltRouteCauseTel2IP 408 Response timeout AltRouteCauseTel2IP 486 User is busy Note This parameter can appear up to 4 times AltRouteCauselP2Tel AltRouteCauselP2Tel lt Call failure reason from Tel gt 5 5 6 For example AltRouteCauselP2Tel 3 No route to destination AltRouteCauselP2Tel 17 Busy here Note This parameter can appear up to 4 times Configuring the Profile Definitions Utilizing the Profiles feature the MediaPack provides high level adaptation when connected to a variety of equipment from both Tel and IP sides and protocols each of which requires a different system behavior Using Profiles use
139. 10 1 21 2 0 11 Major processingErrorAlarm softwareError 46 Board overload alarm An overload condition exists in one or more of the system components Major The overload condition passed Cleared SIP User s Manual 378 Document LTRT 65408 SIP User s Manual F SNMP Traps F 1 2 Component AlarmManager 0 The source varbind text for all the alarms under this component is Board lt n gt AlarmManager 0 where n is the slot number Table F 9 acActiveAlarmTableOverflow Alarm Trap Alarm acActiveAlarmT ableOverflow OID 1 3 6 1 4 15003 9 10 1 21 2 0 12 Default Severity Major Event Type processingErrorAlarm Probable Cause resourceAtOrNearingCapacity 43 Alarm Text Active alarm table overflow Status Changes Condition Too many alarms to fit in the active alarm table Alarm status Major Condition After raise Alarm status Status stays major until reboot A clear trap is not sent Note The status stays major until reboot as it denotes a possible loss of information until the next reboot If an alarm is raised when the table is full it is possible that the alarm is active but does not appear in the active alarm table Corrective Action Some alarm information may have been lost but the ability of the device to perform its basic operations has not been impacted A reboot is the only way to completely clear a problem with the active alarm table Contact your first level group F 1 3 Component EthernetLink 0 The sou
140. 10 10 the Embedded Web Servers Enter Network Password screen appears shown in Figure 5 1 Figure 5 1 Embedded Web Server Login 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 UserName SEmi v Password a V Save this password in your password list Cancel 3 In the User Name and Password fields enter the username default Admin and password default Admin Note that the username and password are case sensitive 4 Click the OK button the Quick Setup screen is accessed shown in Figure 4 1 Version 5 0 51 December 2006 r wi AudioCodes MediaPack 5 3 1 Using Internet Explorer to Access the Embedded Web Server Internet explorers security settings may block access to the gateway s Web browser if they re configured incorrectly In this case the following message is displayed gt To troubleshoot blocked access to Internet Explorer take these 3 steps 1 Delete all cookies from the Temporary Internet files If this does not clear up the problem the security settings may need to be altered continue with Step 2 2 In Internet Explorer gt Tools gt Internet Options e Select the Security tab select Custom Level Scroll down until the Logon options are displayed and change the setting to Prompt for username and password e Select the Advanced tab
141. 10 2 on page 205 and restart the MediaPack the Embedded Web Server uses the provided certificate 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 MediaPack e g 10 3 3 1 instead of a qualified DNS name in the Subject Name This practice 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 Version 5 0 291 December 2006 a WH wt AudioCodes MediaPack 12 2 5 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 MediaPack 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 MediaPack must be configured to use NTP Section 9 7 on page 253 to obtain the current date and time Without a correct date and time c
142. 11 4 on page 357 e Download Status Progress of a TFTP load to a client shown in New IP File shows the IP address applied to the client as a result of the BootP transaction as well as the file name and path of a file transfer for a TFTP transaction Client Name shows the client name as configured for that client in the Client Configuration screen SIP User s Manual 352 Document LTRT 65408 SIP User s Manual C BootP TFTP Configuration Utility Use right click on a line in the Log Window to open a pop up window with the following options Reset Selecting this option results in a reset command being sent to the client VoIP gateway The program searches its database for the MAC address indicated in the line If the client is found in that database the program adds the client MAC address to the Address Resolution Protocol ARP table for the computer The program then sends a reset command to the client This enables a reset to be sent without knowing the current IP address of the client as long as the computer sending the reset is on the same subnet Note To use reset as described above the user must have administrator privileges on the computer Attempting to perform this type of reset without administrator privileges on the computer results in an error message ARP Manipulation Enable must also be turned on in the Preferences window View Client Selecting this option or double clicking on the line in the log windo
143. 112 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 6 3 IP Profile Settings Use the IP Profile Settings screen to define up to four different IP Profiles These Profiles are used in the Tel to IP and IP to Hunt Group Routing tables to associate different Profiles to routing rules IP Profiles can also be used when working with Proxy server set AlwaysUseRouteTable to 1 gt To configure the IP Profile settings take these 9 steps 1 Open the IP Profile Settings screen Protocol Management menu gt Profile Definitions submenu gt IP Profile Settings option the IP Profile Settings screen is displayed Figure 5 24 IP Profile Settings Screen IP Profile Settings Profile ID 1 v Profile Name Profile Parameters Default Ip Profile f Profile Preference 1 v Fax Signaling Method T 38 Relay v Dynamic Jitter Buffer Minimum Delay msec 70 Dynamic Jitter Buffer Optimization Factor 7 RTP IP Diff Serv 164 Signaling DiffServ 184 RTP Redundancy Depth D Remote RTP Base UDP Port 0 CNG Detector Mode Disable Modems Transport Type Enable Bypass NSE Mode Disable Play Ringback Tone to IP Dont Play Enable Early Media Disable Progress Indicator to IP No PI Coder Group
144. 11x FXS amp FXO Series amp MP 124 is in compliance with the essential requirements and other relevant provisions of Directive 89 336 EEC 73 23 EEC 1999 5 ES Estonian K esolevaga kinnitab AudioCodes Ltd seadme MP 11x FXS amp FXO Series amp MP 124 vastavust direktiivi 89 336 EEC 73 23 EEC 1999 5 ES p hin uetele ja nimetatud direktiivist tulenevatele teistele asjakohastele s tetele Finnish AudioCodes Ltd vakuuttaa t ten ett MP 11x FXS amp FXO Series amp MP 124 tyyppinen laite on direktiivin 89 336 EEC 73 23 EEC 1999 5 ES oleellisten vaatimusten ja sit koskevien direktiivin muiden ehtojen mukainen French Par la pr sente AudioCodes Ltd d clare que l appareil MP 11x FXS amp FXO Series amp MP 124 est conforme aux exigences essentielles et aux autres dispositions pertinentes de la directive 89 336 EEC 73 23 EEC 1999 5 ES German Hiermit erkl rt AudioCodes Ltd dass sich dieser diese dieses MP 11x FXS amp FXO Series amp MP 124 in bereinstimmung mit den grundlegenden Anforderungen und den anderen relevanten Vorschriften der Richtlinie 89 336 EEC 73 23 EEC 1999 5 ES befindet Greek ME THN IMAPOY72ZA AudioCodes Ltd AHAQNEI OTI MP 11x FXS amp FXO Series amp MP 124 ZYMMOP ONETAI NPO2 TIZ OYZIOAEIZ ANAITHZEIZ KAI TIX AOINE ZXETIKEZ AIATA EIZ TH OAHMAZ 89 336 EEC 73 23 EEC 1999 5 ES Version 5 0 387 December 2006 7a VT wt AudioCodes MediaPack Hungarian Icelandic I
145. 18 FXS v 4 20 299 410 CSeq 1 REGISTER Contact asip le2 GO IEE A00 Expires 3600 Authorization Digest username 122 realm audiocodes com nonce 11432d6bce58ddf02e3b5elc77c010d2 Wie U0 2 222 response b9c45d0234a5abf5ddf 5c704029b38cf On receiving this request if accepted by the Proxy the proxy returns a 200 OK response closing the REGISTER transaction SIP 2 0 200 OK Wace SiP 2 0 UDE 10 1 1 200 Raomee lt sal ple Ore len OO gt tag lc 2394 0 Tor seias 122010 1 1 2002 Call I1Dy 6549221 94010 1 1 200 Cseq 1 REGISTER Date Thu 26 Jul 2001 09 34 42 GMT Server Columbia SIP Server 1 17 Content Length 0 Contact ssip I 2Z2G10e ie le 2 00 gt mexpires Thuy Zo Jule ZOOM iOS 4426 Mii action proxy q 1 00 Contact lt I22 C0 eZ 00l gt Mexprres iue LOM ane 20si8 0S E407 EEMELI action proxy q 0 00 Expires Thu 26 Jul 2001 10 34 42 GMT 8 12 4 Remote IP Extension between FXO and FXS This application explains how to implement remote extension via IP using 8 port FXO and 8 port FXS MediaPack gateways In this configuration PBX incoming calls are routed to the Remote Extension via the FXO and FXS gateways Requirements mM One FXO MediaPack gateway One FXS MediaPack gateway Analog phones POTS PBX one or more PBX loop start lines E E E m LAN SIP User s Manual 242 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities Con
146. 2 DigitMap for channel O Not Activated lgr_psbrdif 383 0 PSOSBoardInterface PlayTone Called Tone DIAL TONE Short line was detected going to Active Low Code 36010 CID 0 Select the messages copy them and paste them into a text editor such as Notepad Send this txt file to our Technical Support for diagnosis and troubleshooting To clear the screen of messages click on the submenu Message Log the screen is cleared and new messages begin appearing Do not keep the Message Log screen minimized for a prolonged period as a prolonged session may cause the MediaPack to overload As long as the screen is open even if minimized a session is in progress and messages are sent Closing the screen and accessing another stops the messages and terminates the session SIP User s Manual 192 Document LTRT 65408 SIP User s Manual 5 Web Management 5 7 3 Device Information The Device Information screen displays specific hardware and software product information This information can help you to expedite any troubleshooting process Capture the screen and email it to our Technical Support personnel to ensure quick diagnosis and effective corrective action From this screen you can also view and remove any loaded files used by the MediaPack stored in the RAM gt To access the System Information screen m Open the Device Information screen Status amp Diagnostics menu gt Device Information
147. 2 Modem Transport Type that the gateway uses You can select 0 Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 23 Modem Transport Type that the gateway uses You can select 0 Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 32 Modem Transport Type that the gateway uses You can select 0 Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events Note This option applies to V 32 and V 32bis modems V 90 V 34 Modem Transport Type that the gateway uses You can select 0 Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events Number of times that each fax relay payload is retransmitted to the network The valid range is 0 to 2 The default value is 0 Version 5 0 155 December 2006 Ta fa AudioCodes MediaPack Table 5 43 Media Settings Fax Modem CID Parameters continues on pages 154 to 157 Parameter Fax Relay Enhanced Redundancy Depth FaxRelayEnhancedRedundanc yDepth Fax Relay ECM Enable FaxRelayECMEnable Fax Relay Max Rate bps FaxRelayMaxRaite Fax Modem Bypass Coder Type FaxModemBypassCoderType Fax Modem Bypass Packing Factor FaxModemBypassM CNG Detector Mode
148. 2003 EN61000 3 2 2000 A2 2005 EN61000 3 3 1995 A1 2001 EN60950 1 2001 Manufacturer s Name AudioCodes Ltd Manufacturer s Address 1 Hayarden Street Airport City Lod 70151 Israel Type of Equipment Analog VoIP System Model Numbers MP 11x FXS FXO MP 114 2FXS 2FXO Mixed Series MP 118 4FXS 4FXO MP 11x FXS Series MP 112 2FXS MP 114 4FXS MP 118 8FXS MP 11x FXO Series MP 112 2FXO MP 114 4FXO MP 118 8FXO MP 124 FXS Series MP 124D FXS the undersigned hereby declare that the equipment specified above conforms to the above Directives and Standards 27 June 2006 Airport City Lod Israel Signature Date Day Month Year Location I Zusmanovich Compliance Engineering Manager Czech AudioCodes Ltd t mto prohla uje Ze tento MP 11x FXS amp FXO Series amp MP 124 je ve shod se z kladn mi po adavky a dal mi p slu n mi ustanoven mi sm rnice 89 336 EEC 73 23 EEC 1999 5 ES Danish Undertegnede AudioCodes Ltd erkl rer herved at f lgende udstyr MP 11x FXS amp FXO Series amp MP 124 overholder de v sentlige krav og vrige relevante krav i direktiv 89 336 EEC 73 23 EEC 1999 5 ES Dutch Hierbij verklaart AudioCodes Ltd dat het toestel MP 11x FXS amp FXO Series amp MP 124 in overeenstemming is met de essenti le eisen en de andere relevante bepalingen van richtlijn 89 336 EEC 73 23 EEC 1999 5 ES English Hereby AudioCodes Ltd declares that this MP
149. 232 flow control 0 None default 1 Hardware SIP User s Manual 136 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 Advanced Configuration Use this menu to set the gateway s advanced configuration parameters Note Those parameters contained within square brackets are the names used to configure the parameters via the ini file 5 6 1 Configuring the Network Settings From the Network Settings you can Configure the IP Settings refer to Section 5 6 1 1 Configure the Application Settings refer to Section 5 6 1 2 on page 141 Configure the NFS Settings refer to Section 5 6 1 3 on page 143 Configure the IP Routing Table refer to Section 5 6 1 4 on page 145 Configure the VLAN Settings refer to Section 5 6 1 5 on page 147 Configure the PPPoE Settings refer to Section 5 6 1 6 on page 149 Version 5 0 137 December 2006 r wt AudioCodes MediaPack 5 6 1 1 Configuring the IP Settings gt To configure the IP Settings parameters take these 4 steps 1 Open the IP Settings screen Advanced Configuration menu gt Network Settings gt IP Settings option the IP Settings screen is displayed Figure 5 35 IP Settings Screen IP Networking Mode Single IP Network IP Address fi0s82480 Subnet Mask 255 255 0 0 0 Default Gateway Address foso1 o DNS Primary Server IP siz DNS Secondary Server IP DHCP Settings Enable DHCP Di
150. 28 Automatic Dialing Table Screen Automatic Dialing Gateway Port Destination Phone Number Auto Dial Status Port 1 FXS Enable Port 2 FXS Hotline Pot 3 FXS Enable Pot 4 FXS Enable Pot 5 FXO Enable Part6 FXO Enable Port FXO Enable Pot FXO Enable 2 Inthe Destination Phone Number field for a port enter the telephone number to dial 3 From the Auto Dial Status drop down list select one of the following e Enable 1 When a port is selected when making a call the number in the Destination Phone Number field is automatically dialed if phone is offhooked for FXS gateways or ring signal is applied to port FXO gateways e Disable 0 The automatic dialing option on the specific port is disabled the number in the Destination Phone Number field is ignored e Hotline 2 When a phone is offhooked and no digit is pressed for HotLineToneDuration the number in the Destination Phone Number field is automatically dialed applies to FXS and FXO gateways 4 Repeat step 3 for each port you want to use for Automatic Dialing 5 Click the Submit button to save your changes 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Notes e After a ring signal is detected on an Enabled FXO port the gateway initiates a call to the destination numbe
151. 3 6 1 4 1 5003 9 10 1 21 2 0 16 Default Severity Indeterminate Event Type other 0 Probable Cause other 0 Trap Text Keep alive trap Status Changes Condition The STUN client in is enabled and identified a NAT device or doesn t locate the STUN server The ini file contains the following line SendKeepAliveTrap 1 Trap status Trap is sent Note Keep alive is sent every 9 10 of the time defined in the parameter NatBindingDefaultTimeout Table F 12 acPerformanceMonitoringThresholdCrossing Log Trap Trap acPerformanceMonitoringThresholdCrossing OID 1 3 6 1 4 1 5003 9 10 1 21 2 0 27 Default Severity Indeterminate Event Type other 0 Probable Cause other 0 Trap Text Performance Threshold trap was set with source name of performance counter which caused the trap Status Changes Condition A performance counter has crossed the high threshold Trap status Indeterminate Condition A performance counter has crossed the low threshold Trap status cleared Table F 13 acHTTPDownloadResult Log Trap Trap acHTTPDownloadResult OID 1 3 6 1 4 1 5003 9 10 1 21 2 0 28 Default Severity Indeterminate Event Type processingErrorAlarm 3 for failures and other 0 for success Probable Cause other 0 Status Changes Condition Successful HTTP download Trap text HTTP Download successful Condition Failed download Trap text HTTP download failed a network error occurred Note There are other possible textual messag
152. 4 There is a one to one mapping between PBX lines and FXS MediaPack ports Each PBX line is routed to the same phone connected to the FXS MediaPack 5 The call is disconnected when the phone connected to FXS MediaPack goes onhook 8 12 4 3 FXS MediaPack Configuration using the Embedded Web Server gt To configure the FXS MediaPack take these 3 steps 1 In the Endpoint Phone Numbers screen assign the phone numbers 100 to 107 for the gateway s endpoints Endpoint Phone Number Table Channel s Phone Number Hunt Group ID Profile ID 1 FXS 1 8 100 0 2 In the Automatic Dialing screen enter the phone numbers of the FXO MediaPack gateway in the Destination Phone Number fields When a phone connected to port 1 goes offhook the FXS gateway automatically dials the number 200 Automatic Dialing Gateway Port _ Destination Phone Number Auto Dial Status Port 1 FXS 200 Enable Pot2 FXS 201 Enable i Pot 3 FXS 202 Enable Pot4 FXS 203 Enable Pot 5 FXS 204 Enable Port 6 FXS 206 Enable Port 7 FXS 206 Enable Pot 8 FXS 207 Enable SIP User s Manual 244 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 3 In the Tel to IP Routing screen enter 20 in the Destination Phone Prefix field and the
153. 49 Version 5 0 39 December 2006 Ta WH wt AudioCodes MediaPack 3 2 4 1 Connecting the MP 124 RS 232 Port to Your PC Using a standard RS 232 straight cable not a cross over cable with DB 9 connectors connect the MP 124 RS 232 port to either COM1 or COM2 RS 232 communication port on your PC The required connector pinouts and gender are shown below in Figure 3 14 For information on establishing a serial communications link with the MP 124 refer to Section 10 2 on page 262 Figure 3 14 MP 124 RS 232 Cable Wiring DB 9 female for PC DB 9 male for MP 124 SIP User s Manual 40 Document LTRT 65408 SIP User s Manual 4 Getting Started 4 4 1 4 2 Getting Started The MediaPack is supplied with default networking parameters show in Table 4 1 below and with an application software already resident in its flash memory with factory default parameters Before you begin configuring the gateway change its default IP address to correspond with your network environment refer to Section 4 2 and learn about the configuration methods available on the MediaPack refer to Section 4 1 below For information on quickly setting up the MediaPack with basic parameters using a standard Web browser refer to Section 4 2 3 on page 43 Table 4 1 MediaPack Default Networking Parameters FXS or FXO Default Value FXS 10 1 10 10 FXO 10 1 10 11 FXS FXO 10 1 10 10 MediaPack default subnet mask is 255 255 0
154. 58 IP Connectivity Parameters Description IP address defined in the destination IP address field in the Tel to IP Routing table or IP address that is resolved from the host name defined in the destination IP address field in the Tel to IP Routing table Host name or IP address defined in the destination IP address field in the Tel to IP Routing table The method according to which the destination IP address is queried periodically currently only by ping Displays the status of the IP address connectivity according to the method in the Connectivity Method field Can be one of the following 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 Connectivity option is disabled AltRoutingTel2IPMode equals 0 or 2 Determines the QoS according to packet loss and delay of the IP address Can be one of the following Unknown Recent quality information isn t available OK Poor Note 1 This field is applicable only if the parameter AltRoutingTel2IPMode is set to 2 or 3 Note 2 This field is reset if no QoS information is received for 2 minutes Displays QoS information delay and packet loss calculated according to previous calls Note 1 This field is applicable only if the parameter AltRoutingTel2
155. 8 Israel 29 Italy 30 Japan 31 Jordan 32 Kazakhstan 33 Kuwait 34 Latvia 35 Lebanon 36 Luxembourg 37 Macao 38 Malaysia 39 Malta 40 Mexico 41 Morocco 42 Netherlands 43 New_Zealand 44 Nigeria 45 Norway 46 Oman 47 Pakistan 48 Peru 49 Philippines 50 Poland 51 Portugal 52 Romania 53 Russia 54 Saudi_Arabia 55 Singapore 56 Slovakia 57 Slovenia 58 South_Africa 59 South_Korea 60 Spain 61 Sweden 62 Switzerland 63 Syria 64 Taiwan 65 TBR21 66 Thailand 67 UAE 68 United_Kingdom 69 UnitedStates 70 Yemen 71 The default value is 70 United States Disables the generation of Caller ID type 2 when the phone is offhooked 0 Caller ID type 2 isn t played 1 Caller ID type 2 is played default Defines the metering tone 12 kHz or 16 kHz that is detected by FXO gateways and generated by FXS gateways 0 12 kHz metering tone default 1 16 kHz metering tone Note Suitable 12 kHz or 16 KHz coeff must be used for both FXS and FXO gateways Defines the voltage change slope during polarity reversal or wink 0 Soft default 1 Hard Note 1 Some Caller ID signals use reversal polarity and or wink signals In these cases it is recommended to set PolarityReversalType to 1 Hard Note 2 Applicable only to FXS gateways Version 5 0 133 December 2006 Ta WH wt AudioCodes MediaPack Table 5 35 Protocol
156. ADIUS EnableRADIUS RADIUS Accounting Server IP Address RADIUSAccServerlP RADIUS Accounting Port RADIUSAccPort RADIUS Accounting Type RADIUSAccountingType AAA Indications AAAIndications Table 5 32 RADIUS Parameters Description Enables or disables the RADIUS application Valid options include Disables 0 disables RADIUS application default Enable 1 enables RADIUS application 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 Valid options include At Call Release 0 At the release of the call only default At Connect and Release 1 At the connect and release of the call At Setup and Release 2 At the setup and release of the call Determines which Authentication Authorization and Accounting AAA indications to use Valid options include None 0 No indications default Accounting Only 3 Only accounting indications are used SIP User s Manual 126 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 11 Configuring the FXO Parameters Use this screen to configure the gateway s specific FXO parameters gt To configure the FXO parameters take these 4 steps 1 Open the FXO Settings screen Protocol Management menu gt FXO Settings gt FXO Seitings option the FXO Settings screen i
157. Before DID Wink DelayBeforeDIDWink Description Disable 0 DID is disabled default Enable 1 Enable DID If enabled the MediaPack can be used for connection to EIA TIA 464B DID Loop Start lines Both FXO detection and FXS generation are supported An FXO gateway dials DTMF digits after a Wink signal is detected instead of a Dial tone An FXS gateway generates the Wink signal after the detection of offhook instead of playing a Dial tone The time period in seconds after user hangs up the phone and before call is disconnected FXS Also called regret time The default time is 0 seconds Defines the time interval in seconds between detection of offhook and generation of DID Wink Applicable only to FXS gateways The valid range is 0 to 1 000 The default value is 0 Disconnect and Answer Supervision Enable Polarity Reversal EnableReversalPolarity Enable Current Disconnect EnableCurrentDisconnect Disconnect on Broken Connection DisconnectOnBrokenConne ction Broken Connection Timeout BrokenConnectionEventTi meout Disable 0 Disable the polarity reversal service default Enable 1 Enable the polarity reversal service If the polarity reversal service is enabled then the FXS gateway changes the line polarity on call answer and changes it back on call release The FXO gateway sends a 200 OK response when polarity reversal signal is detected applicable to one stage dialing only and
158. CID ETS 300 659 1 Immediate or smooth to prevent erroneous ringing 12 16 KHz sinusoidal bursts Distinctive By frequency 15 100 Hz and cadence patterns Ringing Message Waiting DC voltage generation TIA EIA 464 B V23 FSK data Stutter dial tone and Indication DTMF based Version 5 0 333 December 2006 Ta S3 AudioCodes MediaPack Table 16 1 MP 11x Functional Specifications continues on pages 333 to 335 Voice amp Tone Characteristics Voice Compression Silence Suppression Packet Loss Concealment Echo Canceler Gain Control DTMF Transport in band DTMF Detection and Generation Call Progress Tone Detection and Generation Output Gain Conirol Input Gain Control Fax Modem Relay Fax Relay Modem Transparency Protocols VoIP Signaling Protocol Communication Protocols Line Signaling Protocols Processor Control Processor Control Processor Memory Signal Processors Interfaces FXS Telephony Interface FXO Telephony Interface Combined FXS FXO Network Interface RS 232 Interface Indicators Lifeline G 711 PCM at 64 kbps p law A law msec G 723 1 MP MLQ at 5 3 or 6 3 kbps G 726 at 32 kbps ADPCM msec G 729 CS ACELP 8 Kbps Annex A B G 723 1 Annex A G 729 Annex B PCM and ADPCM Standard Silence Descriptor SID with Proprietary Voice Activity Detection VAD and Comfort Noise Generation CNG G 711 appendix 1 G 723 1 G 729 a b G 165 and G 168 2000 64 msec Programmabl
159. CNGDetectorMode Description 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 Disable 0 Error Correction Mode ECM mode is not used during fax relay Enable 1 ECM mode is used during fax relay default Maximum rate in bps at which fax relay messages are transmitted outgoing calls You can select 2400 0 2 4 kbps 4800 1 4 8 kbps 7200 2 7 2 kbps 9600 3 9 6 kbps 12000 4 12 0 kbps 14400 5 14 4 kbps default Note The rate is negotiated between the sides i e the media server adapts to the capabilities of the remote side Coder the gateway uses when performing fax modem bypass Usually high bit rate coders such as G 711 should be used You can select G 711 A law 64 0 default G 711 u law 1 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 0 Disable default Don t detect CNG 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 2 Events Only CNG is detected on the originating side The CNG signal passes transparently to the remote side and fax session is started Usually T 38 fax session starts when the preamble signal is detected
160. Call Forward table must be defined to use the Call Forward service To define the Call Forward table refer to Section 5 5 9 4 on page 122 Note To use this service the gateways at both ends must support this option No 0 Disable the Call Waiting service Yes 1 Enable the Call Waiting service default If enabled when an FXS gateway receives a call on a busy endpoint it responds with a 182 response and not with a 486 busy The gateway plays a call waiting indication signal When hook flash is detected the gateway switches to the waiting call The gateway that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received Note 1 The gateway s Call Progress Tones file must include a call waiting Ringback tone caller side and a call waiting tone called side FXS only Note 2 The Enable Hold parameter must be enabled on both the calling and the called sides For information on the Call Waiting feature refer to Section 8 1 5 on page 217 For information on the Call Progress Tones file refer to Section 15 1 on page 325 Number of waiting indications that are played to the receiving side of the call FXS only for Call Waiting This parameter is used to control the Registration Subscription rate The default value is 2 SIP User s Manual 84 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 9 Supplementary Service
161. ControlSubnetMask The default subnet mask is 0 0 0 0 Default Gateway Address N A LocalControlDefaultGW Use the IP Routing table instead Advanced Configuration gt Network Settings Media Network Settings available only in Multiple IPs mode IP Address The gateway s source IP address in the Media network LocalMedialPAddress The default value is 0 0 0 0 Subnet Mask The gateway s subnet mask in the Media network LocalMediaSubnetMask The default subnet mask is 0 0 0 0 Default Gateway Address The gateway s default gateway IP address in the Media network LocalMediaDefaultGW The default value is 0 0 0 0 DNS Settings DNS Primary Server IP IP address of the primary DNS server DNSPriServerIP Enter the IP address in dotted format notation for example 10 8 2 255 Note To use Fully Qualified Domain Names FQDN in the Tel to IP Routing table you must define this parameter DNS Secondary Server IP IP address of the second DNS server DNSSecServerlP Enter the IP address in dotted format notation for example 10 8 2 255 Version 5 0 139 December 2006 Ta fal AudioCodes MediaPack Table 5 36 Network Settings IP Settings Parameters continues on pages 138 to 140 Parameter DHCP Settings Enable DHCP DHCPEnable NAT Settings NAT IP Address StaticNatIP Differential Services Description Disable 0 Disable DHCP support on the gateway default Enable 1 Enable DHCP support on
162. Destination Number Prefix b Hunt Group ID c Source Number Prefix d Source IP address obtained from the Contact header in the INVITE message e IP Profile ID Selection of hunt groups for IP to Tel calls is according to destination number source number and source IP address Note 1 To support the in call alternative routing feature users can use two entries that support the same call but assigned it with a different hunt groups The second entree functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauselP2Tel table Note 2 An optional IP ProfilelD 1 to 4 can be applied to each routing rule Note 3 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 wildcard represents any number between 0 and 255 e g 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 Note 4 For available notations that represent multiple numbers refer to Section 5 5 3 1 on page 95 Note 5 This parameter can appear up to 24 times SIP User s Manual 104 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 5 4 Internal DNS Table The internal DNS table similar to a DNS resolution translates hostnames into IP addresses This table is used when hostname translation is required e g Tel to IP Routing table Tw
163. Destination Tel Number 200 Redirect Calling Number Remote Signaling IP 10 8 58 2 Remote RTP IP Port 10 8 58 2 4000 Call Establishment Duration 2 Call Duration 17 Call State SESSION Fax State n a Coder PTime g7231 30 Call Type Voice Call Establishment Method Normal DTMF Selected Method for Tx Rx DTMF_NOT_SUPPORTED 5 8 2 Adding a Port Description The Channel Status screen allows you to add a brief text description or name for each port channel gt To adda port description take these 3 steps 1 Open the Channel Status screen by clicking the Home icon 2 Click a port channel icon and then from the shortcut menu choose Update Port Info a text box appears 7 In the text box type a brief description of this port and then click Apply Port Info 5 8 3 Resetting a Channel The Channel Status screen allows you to inactivate reset a channel This is sometimes useful in cases for example when the gateway FXO is connected to a PBX and the communication between the two can t be disconnected e g using reverse polarity gt To reset channel take these 2 steps 1 Open the Channel Status screen by clicking the Home icon 2 Click a channel icon and then from the shortcut menu choose Reset Channel the channel is changed to inactive SIP User s M
164. 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 MediaPack can be configured to set a different DiffServ value to IP packets according to their class of service Network Premium Media Premium Control Gold and Bronze For the mapping of an application to its class of service refer to Table 9 1 on page 255 The DiffServ parameters are described in Table 5 36 Version 5 0 253 December 2006 a WH wt AudioCodes MediaPack 9 9 VLANS and Multiple IPs Multiple IPs Media Control and Management OAM traffic in the gateway can be assigned one of the following IP addressing schemes E Single IP address for all traffic i e Media Control and OAM E Separate IP address for each traffic type For separate IP addresses the different traffic types are separated into three dedicated networks Instead of a single IP address the gateway is assigned three IP addresses and subnet masks each relating to a different traffic type This architecture enables users to integrate the gateway into a three network environment that is focused on security and segregation Each entity in the gateway e g Web and RTP is mapped to a single traffic type according t
165. Endpoint Phone Number Table click the arrow button next to Endpoint Phone Number For information on how to configure the Endpoint Phone Number Table refer to Section 5 5 7 on page 115 Click the Reset button and then at the prompt click OK the MediaPack applies the changes and restarts You are now ready to start using the VoIP gateway To prevent unauthorized access to the MediaPack it is recommended that you change the username and password that are used to access the Web Interface Refer to Section 5 6 5 1 on page 168 for details on how to change the username and password Once the gateway is configured correctly back up your settings by making a copy of the VoIP gateway configuration ini file and store it in a directory on your PC This saved file can be used to restore configuration settings at a future time For information on backing up and restoring the gateway s configuration refer to Section 5 6 3 on page 165 Version 5 0 47 December 2006 7 T E MediaPack Reader s Notes SIP User s Manual 48 Document LT RT 65408 SIP User s Manual 5 Web Management 5 Web Management The Embedded Web Server is used both for gateway configuration including loading of configuration files and for run time monitoring The Embedded Web Server can be accessed from a standard Web browser such as Microsoft Internet Explorer Netscape Navigator etc Specifically users can employ thi
166. Event Type Probable Cause Alarm Text Note Status Changes Condition Alarm status Condition Alarm status Corrective Action Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status Condition Alarm status Corrective Action Table F 4 acOperationalStateChange Alarm Trap acOperationalStateChange 1 3 6 1 4 1 5003 9 10 1 21 2 0 15 Major processingErrorAlarm outOfService 71 Network element operational state change alarm Operational state is disabled This alarm is raised if the operational state of the node goes to disabled The alarm is cleared when the operational state of the node goes to enabled In IP systems the operational state of the node is disabled if the device fails to properly initialize Operational state changed to disabled Major Operational state changed to enabled Cleared In IP systems check for initialization errors Look for other alarms and Syslogs that might provide additional information about the error Table F 5 acBoardEvResettingBoard Alarm Trap acBoardEvResettingBoard 1 3 6 1 4 1 5003 9 10 1 21 2 0 5 critical equipmentAlarm outOfService 71 User resetting board When a soft reset is triggered via the Web interface or SNMP Critical After raise Status stays critical until reboot A clear trap is not sent A network administrator has taken action to reset the device No corrective action is
167. FXS gateways Sample ini file for MP 114 FXS gateways Sample ini file for MP 112 FXS gateways Default loadable Call Progress Tones dat file Call progress Tones ini file used to create dat file Telephony interface configuration file for MediaPack FXS gateways TrunkPack Downloadable Conversion Utility Syslog server BootP TFTP configuration utility Call Progress Tones Wizard MIB library for SNMP browser Version 5 0 339 December 2006 7 T E Reader s Notes SIP User s Manual 340 Document LTRT 65408 SIP User s Manual B SIP Compliance Tables B SIP Compliance Tables The MediaPack gateways comply with RFC 3261 as shown in the following sections B 1 SIP Functions Table B 1 SIP Functions Function Supported User Agent Client UAC Yes User Agent Server UAS Yes Proxy Server Third party only tested with amongst others Ubiquity Delta3 Microsoft 3Com BroadSoft Snom and Cisco Proxies Redirect Server Third party Registrar Server Third party Event Publication Agent Yes EPA Event State Compositor Third party ESC B 2 SIP Methods Table B 2 SIP Methods Method Supported Comments INVITE Yes ACK Yes BYE Yes CANCEL Yes REGISTER Yes Send only REFER Yes NOTIFY Yes INFO Yes OPTIONS Yes PRACK Yes UPDATE Yes PUBLISH Yes Send only SUBSCRIBE Yes Version 5 0 341 December 2006 Ta r 8 AudioCodes MediaPack B 3 Header Field
168. H wt AudioCodes MediaPack 15 2 15 2 1 Prerecorded Tones PRT File The Call Progress Tones mechanism has several limitations such as a limited number of predefined tones and a limited number of frequency integrations in one tone To work around these limitations and provide tone generation capability that is more flexible the PRT file can be used If a specific prerecorded tone exists in the PRT file it takes precedence over the same tone that exists in the CPT file and is played instead of it Note that the prerecorded tones are used only for generation of tones Detection of tones is performed according to the CPT file PRT File Format The PRT dat file contains a set of prerecorded tones to be played by the MediaPack during operation Up to 40 tones totaling approximately one minute can be stored in a single file in flash memory The prerecorded tones raw data PCM or L8 files are prepared offline using standard recording utilities such as CoolEdit and combined into a single file using the TrunkPack Downloadable Conversion utility refer to Section E 1 3 on page 366 The raw data files must be recorded with the following characteristics E Coders G 711 A law G 711 law or Linear PCM E Rate 8 kHz E Resolution 8 bit E Channels mono The generated PRT file can then be loaded to the MediaPack using the BootP TFTP utility refer to Section 5 9 2 1 on page 203 or via the Embedded Web Server Section 5 9 2 on pag
169. I P User s Manual 234 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 10 1 RADIUS Server Messages In Figure 8 5 below non standard parameters are preceded with brackets Figure 8 5 Accounting Example Accounting Request 361 user name 111 acct session id 1 nas ip address 2a ine oe oS nas port type 0 acct status type 2 acct input octets 4841 acct output octets 8800 acct session time 1 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 3f d61009 Oe2f3cc5 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 Version 5 0 235 December 2006 Ta WH wt AudioCodes MediaPack 8 11 Proxy or Registrar Registration Example The REGISTER message is sent to the Registrars IP address if configured or to the Proxy s IP address The message is sent per gateway or per gateway endpoint according to the AuthenticationMode parameter Usually the FXS gateways are registered per gateway port while FXO gateways send a single registration message where Username is used instead of phone number in From To head
170. IP gateway configuration parameters and normally appears as ini Timeout This specifies the number of seconds that the TFTP utility waits before retransmitting TFTP messages This can be left at the default value of 5 the more congested your network the higher the value you should define in these fields Maximum Retransmissions Specifies the number of times that the TFTP utility tries to resend messages after timing out This can be left at the default value of 10 the more congested your network the higher the value should be defined in these fields SIP User s Manual 354 Document LTRT 65408 SIP User s Manual C BootP TFTP Configuration Utility C 11 Configuring the BootP Clients The Clients window shown in Figure C 4 is used to set up the parameters for each specific VoIP gateway Figure C 4 Client Configuration Screen Client Configuration x Client MAC 00 30 8F 64 64 1 2 v BY 00 90 8F 10 22 33 10 8 201 120 i EF 00 90 8F 55 42 21 10 8 201 1 Client Name E7 00 90 9F 64 64 12 10 8 201 10 Template lt none gt IP po a ae Subnet 255 255 fo fo a Gateway fio fe fo fi f TFTP Server IP fio fe fi 21 fal Boot File erx cmp v ia INI File in X fe Apply Apply amp Reset C 11 1 Adding Clients Adding a client creates an entry in the BootP Tool for a specific gateway gt To add a client to the list without using a template take these 3 steps Click Add New Client gy a client wi
171. IPMode is set to 2 or 3 Note 2 This field is reset if no QoS information is received for 2 minutes Can be one of the following DNS Disable DNS Resolved DNS Unresolved SIP User s Manual 188 Document LTRT 65408 SIP User s Manual 5 Web Management 5 7 1 2 Call Counters Number of Attempted Calls Number of Established Calls The Call Counters screens provide you with statistic information on incoming IP gt Tel and outgoing Tel gt IP calls The statistic 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 CDR message is sent The release reason can be viewed in the Termination Reason field in the CDR message For detailed information on each counter refer to Table 5 59 on page 189 You can reset this information by clicking the Reset Counters button gt To view the IP gt Tel and Tel gt IP Call Counters information take this step H Open the Call Counters screen you want to view Status amp Diagnostics menu gt Gateway Statistics submenu the relevant Call Counters screen is displayed Figure 5 56 shows the Tel gt IP Call Counters screen Figure 5 56 Tel gt IP Call Counters Screen Tel to IP Calls Count Number of Attempted Calls 10 Number of Established Calls 5 Percentage of Successful Calls 50 000000 Number of Failed Calls due to a Busy Line Number of Failed Calls due
172. IPSec and IKE on the gateway set the ini file parameter EnablelPSec to 1 12 1 3 1 IKE Configuration The parameters described in Table 12 1 below are used to configure the first phase main mode of the IKE negotiation for a specific peer A different set of parameters can be configured for each of the 20 available peers Up to two IKE main mode proposals Encryption Authentication DH group combinations can be defined The same proposals must be configured for all peers Table 12 1 IKE Table Configuration Parameters continues on pages 281 on 282 Parameter Name Shared Key IKEPolicySharedKey First to Fourth Proposal Encryption Type IKEPolicyProposalEncryptio n_X First to Fourth Proposal Authentication Type IKEPolicyProposalAuthentic ation_X First to Fourth Proposal DH Group IKEPolicyProposalDHGroup _X Description Determines the pre shared key in textual format Both peers must register the same pre shared key for the authentication process to succeed Note 1 The pre shared key forms the basis of IPSec security and should therefore be handled cautiously in the same way as sensitive passwords It is not recommended to use the same pre shared key for several connections Note 2 Since the inifile is in plain text format loading it to the gateway over a secure network connection is recommended preferably over a direct crossed cable connection from a management PC For added confidentiality use th
173. LAN Settings parameters take these 4 steps Open the VLAN Settings screen Advanced Configuration menu gt Network Settings gt VLAN Settings option the VLAN Settings screen is displayed Figure 5 39 VLAN Settings Screen VLAN Settings VLAN Mode ID Settings Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID Priority Settings Network Priority Media Premium Priority Control Premium Priority 2 Configure the VLAN Settings according to Table 5 40 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Version 5 0 147 December 2006 Ta fal AudioCodes MediaPack Table 5 40 Network Settings VLAN Settings Parameters Parameter VLAN Mode VianMode IP Settings Native VLAN ID VlanNativeVlanID OAM VLAN ID VianOamVlanID Control VLAN ID VlanControlVianID Media VLAN ID VlanMediaVlanID Priority Settings Network Priority VlanNetworkServiceClassPriority Media Premium Priority VianPremiumServiceClassMediaPrior ity Control Premium Priority VianPremiumServiceClassControlIPri ority Gold Priority VlanGoldServiceClassPriority Bronze Priority VlanBronzeServiceClassPriority ini File Parameters EnableDNSasOAM EnableNTPasOAM VlanSendNonTaggedOnNative Description Sets the VLAN functionality Disable 0 default Enable
174. MAT SNMPUsers_Index SNMPUsers_Username SNMPUsers_AuthProtocol SNMPUsers_PrivProtocol SNMPUsers_AuthKey SNMPUsers_PrivKey SNMPUsers_Group SNMPUsers 0 w3user 0 0 0 SNMPUsers 1 v3adminl 1 0 myauthkey 1 SNMPUsers 2 v3admin2 2 1 myauthkey myprivkey 1 SNMPUsers 14 8 2 2 The example above creates the following three v3 users m The user v3user is defined for a security level of noAuthNoPriv 1 and is associated with ReadGroup1 m The user v3admin1 is defined for a security level of authNoPriv 2 with authentication protocol MD5 The authentication text password is myauthkey and the user will be associated with ReadWriteGroup2 m The user v3admin2 is defined for a security level of authPriv 3 with authentication protocol SHA 1 and privacy protocol DES The authentication text password is myauthkey the privacy text password is myprivkey and the user will be associated with ReadWriteGroup3 Configuring SNMP v3 Users via SNMP To configure SNMP v3 users the EM must use the standard snmpUsmMIB and the snmpVacmMIB gt To adda read only noAuthNoPriv SNMPv3 user v3user take these 3 steps 1 Clone the row with the same security level After the clone step the status of the row is notReady 3 2 Activate the row i e set the row status to active 1 9 Add a row to the vacmSecurityToGroupTable for SecurityName v3user GroupName ReadGroup1 and Security
175. Management gt To configure the Generate Caller ID to Tel Table take these 5 steps 1 Open the Generate Caller ID to Tel screen Protocol Management menu gt Endpoint Settings gt Generate Caller ID to Tel option the Generate Caller ID to Tel screen is displayed Figure 5 30 MediaPack FXS Generate Caller ID to Tel Screen Generate Caller ID to Tel Port 1 Disable bi Port 2 Enable Port 3 Port 4 Port 6 Port 7 Port 8 Port 5 2 In the Caller ID field 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 Empty Caller ID generation FXS or detection FXO for the specific port is determined according to the parameter EnableCallerID described in Table 5 9 3 Repeat step 2 for each port Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 A Table 5 30 Generate Caller ID to Tel ini File Parameter Parameter Name in ini File Parameter Format EnableCallerID_X EnableCallerID_ lt Port gt lt Caller ID gt Caller ID 0 Disabled default 1 Enabled If not configured use the global parameter EnableCaller D Note 1 The numbering of ports starts with 0 Note 2 This parameter c
176. Management ini File Parameters continues on pages 132 to 137 ini File Parameter Name CurrentDisconnectDurat ion CurrentDisconnectDefa ultThreshold TimeToSampleAnalogLi neVoltage AnalogCallerIDTimimgM ode EnableRAIl RAIHighThreshold RAlILowThreshold RAILoopTime Valid Range and Description Duration of the current disconnect pulse in msec The default is 900 msec The range is 200 to 1500 msec Applicable for both FXS and FXO gateways Note The FXO gateways detection range is 200 msec of the parameter s value 100 For example if CurrentDisconnectDuration 200 the detection range is 100 to 500 msec Determines the line voltage threshold which when reached is considered a current disconnect detection Note Applicable only to FXO gateways The valid range is 0 to 20 Volts The default value is 4 Volts Determines the frequency at which the analog line voltage is sampled after offhook for detection of the current disconnect threshold Note Applicable only to FXO gateways The valid range is 100 to 2500 msec The default value is 1000 msec 0 Caller ID is generated between the first two rings default 1 The gateway attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type Note that when used with distinctive ringing the Caller ID signal doesn t change the distinctive ringing timing Note Applicable only to FXS gateways 0 Disable
177. Manual 8 Telephony Capabilities 8 4 2 4 FXO Supplementary Services Hold Transfer toward the Tel side The ini file parameter LineTransferMode must be set to 0 default If the FXO receives a hook flash from the IP side using out of band or RFC 2833 the gateway sends the hook flash to the Tel side by 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 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 gateway 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 8 5 ThroughPacket The gateway supports a proprietary
178. MediaPack Table 5 18 IP to Hunt Group Routing Table continues on pages 103 to 104 Parameter Source IP Address Description Each entry in the Source IP Address fields represents the source IP address of an IP gt Tel call obtained from the Contact header in the INVITE message Note The source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 Any SIP incoming call whose destination number matches the prefix defined in the Destination Phone Prefix field and its source number matches the prefix defined in the adjacent Source Phone Prefix field and its source IP address matches the address defined in the Source IP Address field is assigned to the hunt group entered in the field to the right of these fields Note that IP to hunt group routing can be performed according to any combination of source destination phone prefixes and source IP address or using each independently Note For available notations that represent multiple numbers used in the prefix columns refer to Section 5 5 3 1 on page 95 Hunt Group ID Profile ID Parameter Name in ini File PSTNPrefix In each of the Hunt Group ID fields enter the hunt group ID to which calls that match these prefixes are assigned Enter the number of the IP profile that is assigned to the routing rule Parameter Format PSTNPrefix a b c d e a
179. Model usm 3 Note A row with the same security level noAuthNoPriv must already exist in the usmUserTable See the usmUserTable for details SIP User s Manual 316 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 1 2 3 4 2 3 To delete the read only noAuthNoPriv SNMPv3 user v3user take these 3 steps If v3 user is associated with a trap destination follow the procedure for associating a different user to that trap destination See below Delete the vacmSecurityToGroupTable row for SecurityName v3user GroupName ReadGroup1 and SecurityModel usm Delete the row in the usmUserTable for v3user To add a read write authPriv SNMPv3 user v3user take these 4 steps Clone the row with the same security level Change the authentication key and privacy key Activate the row That is set the row status to active 1 Add a row to the vacmSecurityToGroupTable for SecurityName v3admin1 GroupName ReadWriteGroup3 and SecurityModel usm 3 Note A row with the same security level authPriv must already exist in the usmUserTable see the usmUserTable for details To delete the read write authPriv SNMPv3 user v3admin1 take these 3 steps If v3admin1 is associated with a trap destination follow the procedure for associating a different user to that trap destination See below Delete the vacmSecurityToGroupTable row for SecurityName v3admin1 GroupName ReadWriteG
180. P IP v6 Control Protocol and BCP PPP Bridging Control Protocol Each of them handles and manages the specific needs required by their respective network layer protocol When working in an IP network IPCP is used as the Network Configuration Protocol The IPCP is used to configure the network layer of the hosts requesting declaring on IP Addresses Further information on PPP Protocol is available on the IETF website http www ietf org rfc rfc1661 txt Further information on Password Authentication Protocol is available on the IETF website http www ietf org rfc rfc1334 txt Further information on Challenge Handshake Authentication Protocol is available on the IETF website htto www ietf org rfc rfc1994 txt Further information on PPP Internet Protocol Control Protocol IPCP is available on the IETF website http www ietf org rfc rfc1332 txt SIP User s Manual 250 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities 9 4 2 9 4 3 PPPoE Overview PPPoE is a method of sending the Point to Point Protocol over Ethernet network PPPoE provides the ability to connect a network of hosts over a simple bridging access device to a remote Access Concentrator Access control billing and type of service can be done on a per user rather than a per site basis A common use of the PPPoE is in the ADSL market The home PC is connected to a modem via Ethernet and the PC uses the PPPoE to simulate as if i
181. PEnable described in Table 5 16 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 Version 5 0 187 December 2006 Ta A wi AudioCodes MediaPack gt To view the IP connectivity information take these 2 steps N Set AltRoutingTel2IPEnable to 1 or 2 Open the IP Connectivity screen Status amp Diagnostics menu gt Gateway Statistics submenu gt IP Connectivity the IP Connectivity screen is displayed Figure 5 55 IP Connectivity IP Address Host Name Figure 5 55 IP Connectivity Screen Connectivity Connectivity Method Status Quality Status Quality Info DNS Status 10 13 77 7 10 13 77 7 Ping CON_OK Q03_UNKNOWN PL percent 0 DELAY msec 0 DNS_DISABLE 10 13 77 9 10 13 77 9 Ping CON_OK QOS_UNKNOWN PL percent 0 DELAY msec 0 DNS_DISABLE 10 13 77 18 10137718 Ping CON_FAIL QOS_UNKNOWN PL percent 0 DELAY msec 0 DNS_DISABLE 1 2 3 4 doron_pe Ping CON FAIL OS_UNKNOWN PL percent 0 DELAY msec 0 DNS_RESOLVED 10 13 2 95 XYZ PL percent 0 DELAY Ping CON_INIT QOS_UNKNOWN 7 msec 0 DNS_UNRESOLVED UNUSED ENTRY UNUSED ENTRY Column Name IP Address Host Name Connectivity Method Connectivity Status Quality Status Quality Info DNS Status Table 5
182. Pv3 user name Figure 14 1 presents an example of entries in a device ini file regarding SNMP The device can be configured to send to multiple trap destinations The lines in the file below are commented out with the 6 at the beginning of the line All of the lines below are commented out since the first line character is a semi colon Figure 14 1 Example of Entries in a Device ini file Regarding SNMP SNMP trap destinations The board maintains a table of trap destinations containing 5 rows The rows are numbered 0 4 Each block of 4 items below apply to a row in the table To configure one of the rows uncomment all 4 lines in that block IP address and if necessary change the port number set ISUSED to 0 change these entries as needed SNMPManagerTableIP_0 SNMPManagerTrapPort_0 162 sUsed_0 1 SNMPManagerTrapSendingEnable_0 1 Supply an To delete a trap destination 7 SNMPManager SNMPManagerTableIP_1 SNMPManagerTrapPort_1 162 sUsgecl I SNMPManagerTrapSendingEnable_1 1 SNMPManager i SNMPManagerTableIP_2 SNMPManagerTrapPort_2 162 sUsed_2 1 SNMPManagerTrapSendingEnable_2 1 7 SNMPManager i SNMPManagerTableIP_3 SNMPManagerTrapPort_3 162 sUsed_3 1 SNMPManagerTrapSendingEnable_3 1 SNMPManager SNNMPManagerTableIP_4 SNMPManagerTrapPort_4 162 sUsed_4 1 SNMPManagerTrapSendingEnable_4 1 SNMPManager To
183. R Report Level Debug Level Misc Parameters Progress Indicator to IP Enable X Channel Header Not Configured Disable Enable Busy Out Disable Default Release Cause Delay After Reset sec Max Number of Active Calls Max Call Duration min Enable LAN Watchdog Enable Calls Cut Through Disable Disable Enable User Information Usage Disable Out Of Service Behavior Reorder Tone Version 5 0 77 December 2006 Ta WH wt AudioCodes MediaPack 2 Configure the general parameters under Advanced Parameters according to Table 5 8 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 a Table 5 8 Advanced Parameters General Parameters continues on pages 78 to 82 Parameter Description Signaling DiffServ ControllPDiffServ IP Security No 0 Gateway accepts all SIP calls default SecureCallsFromIP Yes 1 Gateway accepts SIP calls only from IP addresses defined in the Tel to IP routing table The gateway rejects all calls from unknown IP addresses For detailed information on the Tel to IP Routing table refer to Section 5 5 5 2 on page 100 Obsolete parameter use PremiumServiceClassControlDiffServ instead Note Specifying the IP address of a Proxy server in the Tel to IP R
184. Response code Upon receiving the final success or failure response the gateway 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 UAS Behavior History Info is sent in the final response only Upon receiving a request with History Info the UAS checks the policy in the request If session header or history policy tag is found the final response is sent without History Info Otherwise it is copied from the request Applicable to Tel gt IP calls No 0 The Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name if Tel Display Name is received If no Display Name is received from the Tel side the IP Display Name remains empty default Yes 1 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 Te
185. SI 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 Selects the Bellcore VMWI sub standard 0 Between rings default 1 Not ring related Version 5 0 163 December 2006 Ta fal AudioCodes MediaPack Table 5 47 Media Settings ini File Parameters continues on pages 162 to 163 ini File Parameter Name StunServerDomainName NATBindingDefaultTimeout TxDTMFHangOverTime RxDTMFHangOverTime Valid Range and Description Defines the domain name for the 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 Note You can either use the STUNServerPrimarylP or the STUNServerDomainName parameter with priority to the former one Defines the NAT binding lifetime in seconds STUN refreshes the binding information after this time expires The valid range is 0 to 2592000 The default value is 30 Defines the Voice Silence time in msec units after detecting the end of DTMF or MF digits at the Tel PSTN side when the DTMF Transport Type
186. Section 3 2 4 1 on page 40 For the MP 11x refer to Section 3 1 5 1 on page 34 2 Use serial communication software e g HyperTerminal to connect to the MediaPack Set your serial communication software to the following communications port settings Baud Rate 115 200 bps MP 124 9 600 bps MP 11x Data bits 8 Parity None Stop bits 1 Flow control None The CLI prompt becomes available SIP User s Manual 44 Document LTRT 65408 SIP User s Manual 4 Getting Started 4 2 4 2 Assign an IP Address gt 1 2 To assign an IP address via the CLI take these 4 steps At the prompt type conf and press enter the configuration folder is accessed To check the current network parameters at the prompt type GCP IP and then press Enter the current network settings are displayed Change the network settings by typing SCP IP ip address subnet_mask default_gateway e g SCP IP 10 13 77 7 255 255 0 0 10 13 0 1 the new settings take effect on the fly Connectivity is active at the new IP address Note This command requires you to enter all three network parameters each separated by a space To save the configuration at the prompt type SAR and then press Enter the MediaPack restarts with the new network settings Version 5 0 45 December 2006 Ta Ca AudioCodes MediaPack 4 3 Configure the MediaPack Basic Parameters To configure the MediaPack basic parameters us
187. TE message Use the Waiting For Dial Tone parameter to specify whether the dialing should come after detection of dial tone or immediately after seizing of the line Version 5 0 127 December 2006 Ta WH wt AudioCodes MediaPack Table 5 33 FXO Parameters continues on pages 127 to 130 Parameter Waiting For Dial Tone IsWaitForDialTone Time to Wait before Dialing msec WaitForDialTime Note Replaces the obsolete parameter FXOWaitForDialTime Ring Detection Timeout sec FXOBetweenRingTime Description No 0 Don t wait for dial tone Yes 1 Wait for dial tone default Used for IP gt MediaPack FXO gateways when One Stage Dialing is enabled If wait for dial tone is enabled the FXO gateway dials the phone number to the PSTN PBX line only after it detects a dial tone Note 1 The correct dial tone parameters should be configured in the Call Progress Tones file Note 2 It can take the gateway 1 to 3 seconds to detect a dial tone according to the dial tone configuration in the Call Progress Tones file If Waiting For Dial Tone is disabled the FXO gateway immediately dials the phone number after seizing the PSTN PBX line without listening to dial tone Determines the delay before the gateway starts dialing on the FXO line in the following scenarios applicable only to FXO gateways 1 The delay between the time the line is seized and dialing is begun dur
188. Table 10 4 according to the procedure described in Section 10 6 4 on page 276 m To replace AudioCodes default logo with a text string via the ini file add modify the two ini file parameters in Table 10 4 according to the procedure described in Section 6 2 on page 209 Table 10 4 Web Appearance Customizable ini File Parameters Parameter Description UseWebLogo 0 Logo image is used default 1 Text string is used instead of a logo image WebLogoText Text string that replaces the logo image The string can be up to 15 characters Version 5 0 273 December 2006 Ta WH wt AudioCodes MediaPack 10 6 2 Replacing the Background Image File Parameter The background image file is duplicated across the width of the screen The number of times the image is duplicated depends on the width of the background image and screen resolution When choosing your background image keep this in mind Note Use a gif jpg or jpeg file for the background image It is important that the image file has a fixed height of 59 pixels The size of the image files logo and background is limited each to 64 kbytes To replace the background image via the Web take these 6 steps Access the MediaPack Embedded Web Server refer to Section 5 3 on page 51 In the URL field append the suffix AdminPage note that it s case sensitive to the IP address e g http 10 1 229 17 AdminPage Click the Image Load to Device the Image
189. The BootP parameters are special Hidden parameters Once defined and saved in the flash memory they are used even if they don t appear in the ini file BootPRetries This parameter is used to Note This parameter only takes effect from the next reset of the gateway Set the number of BootP requests Set the number of DHCP packets the gateway sends during start up the gateway sends The gateway stops sending BootP After all packets were sent if requests when either BootP reply is there s still no reply the gateway received or number of retries is loads from flash reached 1 4 DHCP packets 1 1 BootP retry 1 second 2 5 DHCP packets 2 2 BootP retries 3 second 3 6 DHCP packets default 3 3 BootP retries 6 second 4 7 DHCP packets default 5 8 DHCP packets 4 10 BootP retries 30 second 6 9 DHCP packets 5 20 BootP retries 60 second 7 10 DHCP packets 6 40 BoofP retries 120 second 15 18 DHCP packets 7 100 BootP retries 300 second 15 BootP retries indefinitely Version 5 0 183 December 2006 Ta fal AudioCodes MediaPack Table 5 55 Board ini File Parameters continues on pages 182 to 184 ini File Parameter Name BootPSelectiveEnable BootPDelay ExtBootPReqEnable Valid Range and Description Enables the Selective BootP mechanism 1 Enabled 0 Disabled default The Selective BootP mechanism available from Boot version 1 92 enables the gateway
190. ToLog AFL Device Reset ActivityListToLog DR Flash Memory Burning ActivityListToLog FB Device Software Update ActivityListToLog SWU Changes made on the fly to parameters Loading of auxiliary files e g via Certificate screen Device reset via the Maintenance Actions screen Burning of files parameters to flash e g Maintenance Actions screen cmp loading via the Software Upgrade Wizard Version 5 0 177 December 2006 Ta Ce AudioCodes MediaPack Table 5 51 Management Settings Parameters continues on pages 177 to 178 Parameter Access to Restricted Domains ActivityListToLog ARD Non Authorized Access ActivityListToLog NAA Sensitive Parameters Value Change ActivityListToLog SPC ini file example Description Access to Restricted Domains The following screens are restricted 1 ini parameters AdminPage 2 General Security Settings 3 Configuration File 4 IPSec IKE tables 5 Software Upgrade Key 6 Internal Firewall 7 Web Access List 8 Web User Accounts Attempt to access the Embedded Web Server with a false empty username or password Changes made to sensitive parameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 4 ActivityListToLog ActivityListToLog pvc afl dr fb swu ard naa spc 5 6 6 1 Configuring the SNMP Managers Table The SNMP Managers table allows you to c
191. Transport Type V22ModemTransportType V 23 Modem Transport Type V23ModemTransportType V 32 Modem Transport Type V32ModemTransportType V 34 Modem Transport Type V34ModemTransportType Fax Relay Redundancy Depth FaxRelayRedundancyDepth Description Defines one of the following standards for detection FXO and generation FXS of Caller ID and detection FXO of MWI when specified signals Bellcore 0 Caller ID and MWI default ETSI 1 Caller ID and MWI NTT 2 Backward Compatible 3 British 4 DTMF ETSI 16 Denmark 17 Caller ID and MWI India 18 Brazil 19 Note 1 The Caller ID signals are generated detected between the first and the second rings Note 2 To select the Bellcore Caller ID sub standard use the parameter BellcoreCalleriIDTypeOneSubStandard To select the ETSI Caller ID sub standard use the parameter ETSICalleriIDTypeOneSubStandard Note 3 To select the Bellcore MWI sub standard use the parameter BellcoreVMWITypeOneStandard To select the ETSI MWI sub standard use the parameter ETSIVMWITypeOneStandard Note 4 If you define NTT i e 2 for the caller ID type you need to define the NTT DID signaling form FSK or DTMF using NTTDIDSignallingForm V 21 Modem Transport Type that the gateway uses You can select 0 Disable Transparent default 1 Enable Relay N A 2 Enable Bypass 3 Events Only Transparent with Events V 2
192. UDP port used to receive SIP messages The valid range is 1 to 65534 The default value is 5060 Local TCP port used to receive SIP messages The default value is 5060 Local TLS port used to receive SIP messages The default value is 5061 Note The value of TLSLocalSIPPort must be different to the value of TCPLocalSIPPort Enables secured SIP SIPS connections over multiple hops Disable 0 default Enable 1 When SIPTransportType 2 TLS and EnableSIPS is disabled TLS is used for the next network hop only When SIPTransportType 2 TLS or 1 TCP and EnableSIPS is enabled TLS is used through the entire connection over multiple hops Note If SIPS is enabled and SIPTransportType UDP the connection fails Enables the reuse of the same TCP connection for all calls to the same destination Valid options include 0 Use a separate TCP connection for each call default 1 Use the same TCP connection for all calls 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 No 0 user phone string isn t used in SIP URI Yes 1 user phone string is part of the SIP URI default No 0 Doesn t use user phone string in From header default Yes 1 juser phone string is part of the From header Determines the format of the URI in the P A
193. US authentication is used the gateway doesn t store the username and password but simply forwards them to the pre configured RADIUS server for authentication acceptance or rejection The internal Web Telnet passwords can be used as a fallback mechanism in case the RADIUS server doesn t respond configured by the parameter BehaviorUponRadiusTimeout Note that when RADIUS authentication is performed the Web Telnet servers are blocked until a response is received with a timeout of 5 seconds RADIUS authentication requires HTTP basic authentication meaning the username and password are transmitted in clear text over the network Therefore users are recommended to set the parameter HttpsOnly 1 to force the use of HTTPS since the transport is encrypted Setting Up a RADIUS Server The following examples refer to FreeRADIUS a free RADIUS server that can be downloaded from www freeradius org Follow the directions on that site for information on installing and configuring the server If you use a RADIUS server from a different vendor refer to its appropriate documentation gt To set up a RADIUS server take these 5 steps 1 Define the gateway as an authorized client of the RADIUS server with a predefined shared secret a password used to secure communication and a vendor ID The figure below displays an example of the file clients conf FreeRADIUS client configuration Figure 12 12 Example of the File clients c
194. USM users m SNMP encoded over IPSec refer to Section 12 1 on page 279 Combinations of the above Currently both SNMP and ini file commands and downloads are not encrypted For ini file encoding refer to Section E 1 2 on page 365 SNMP Community Names By default the device uses a single read only community string of public and a single read write community string of private The following community strings can be defined m Up to five read only community strings E Up to five read write community strings E A single trap community string Each community string must be associated with one of the following predefined SNMP groups Table 14 2 SNMP Predefined Groups Group Gets Access Sets Access Sends Traps ReadGroup Yes No Yes ReadWriteGroup Yes Yes Yes TrapGroup No No Yes 14 8 1 1 Configuration of Community Strings via the Web For detailed information on configuration the community strings via the Embedded Web Server refer to Section 5 6 6 2 on page 180 14 8 1 2 Configuration of Community Strings via the ini File The following ini file parameters are used to configure community strings E SNMPReadOnlyCommunityString_ lt x gt HHHH E SNMPReadWriteCommunityString_ lt x gt HHHHHH Where lt x gt is a number from 0 to 4 The character represents any alphanumeric character The maximum length of the string is 20 characters Version 5 0 313 December 2006 7a Ta wi AudioCodes MediaPack
195. Usm MIB this MIB RFC 3414 implements the user based Security Model SNMP Vacm MIB This MIB RFC 3415 implements the view based Access Control Model SNMP Community MIB This MIB RFC 3584 implements community string management E ipForward MIB RFC 2096 fully supported Version 5 0 309 December 2006 a WH wt AudioCodes MediaPack RTCP XR This MIB RFC implements the following partial support e The rtcpXrCallQualityTable is fully supported e Inthe rtcpXrHistoryTable support of the RCQ objects is provided only with no more than 3 intervals 15 minutes long each e Supports the rtcpXrVoipThresholdViolation trap In addition to the standard MIBs the complete product series contains proprietary MIBs AC TYPES MIB lists the known types defined by the complete product series This is referred to by the sysObjectID object in the MIB II Note The acBoard MIB is still supported but is being replaced by five newer proprietary MIBs As noted above five new MIBs cover the device s general parameters Each contains a Configuration subtree for configuring related parameters In some there also are Status and Action subtrees The 5 MIBs are e AC ANALOG MIB e AC CONTROL MIB e AC MEDIA MIB e AC PSTN MIB e AC SYSTEM MIB Other proprietary MIBs are acGateway MIB This proprietary MIB contains objects related to configuration of the device when applied as a SIP or H 323 media gatewa
196. WH wt AudioCodes MediaPack F5 10 8 201 161 gt 10 8 201 158 BYE BYE sip 2000 10 8 201 158 user phone user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 161 branch z9hG4bKacLBzZgmA From lt sip 6000 10 8 201 161 gt tag 1c29715 ios lt sip 2000CTO Ns 20 le lS3 gt tag lc3535 Call ID 2123353775377NrpL 2000 6000 10 8 201 158 User Agent Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 CSeq 34541 BYE Supported 100rel em Content Length 0 F6 10 8 201 158 gt 10 8 201 161 200 OK SEY 2a Ome OO MOK Via SIP 2 0 UDP 10 8 201 161 branch z9hG4bKacLBzZgmA From lt sip 6000 10 8 201 161 gt tag 1c29715 To lt sup Z2O00CTORse 20 aS ee yitag le sso Call ID 212335377537 7NrpL 2000 6000 10 8 201 158 Server Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 CSeq 34541 BYE Supported 100rel em Content Length 0 8 12 3 SIP Authentication Example MediaPack gateways support 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 resend the INVITE with a Proxy Authorization header containing the credential
197. XXXxX Xx 9 1 XXXXXXXXXX 901 1x T Dial Tone Duration sec Time in seconds that the dial tone is played TimeForDialTone The default time is 16 seconds FXS gateway ports play the dial tone after phone is picked up while FXO gateway ports play the dial tone after port is seized in response to ringing Note 1 During play of dial tone the gateway waits for DTMF digits Note 2 TimeForDialTone is not applicable when Automatic Dialing is enabled Hotline Dial Tone Duration Duration in seconds of the Hotline dial tone HotLineToneDuration If no digits are received during the Hotline dial tone duration the gateway initiates a Call to a preconfigured number set in the automatic dialing table The valid range is 0 to 60 The default time is 16 seconds Applicable to FXS and FXO gateways Enable Special Digits Disable 0 or terminate number collection default IsSpecialDigits Enable 1 if you want to allow and to be used for telephone numbers dialed by a user or entered for the endpoint telephone number Note The and can always be used as first digit of a dialed number even if you select Disable for this parameter Default Destination Number Defines the telephone number that the gateway uses if the parameter DefaultNumber TrunkGroup_x doesn t include a phone number The parameter is used as a starting number for the list of channels comprising all hunt groups in the gateway 5 5
198. a DiffServ Control Premium QoS PremiumServiceClassContr olDiffServ Gold QoS GoldServiceClassDiffServ Bronze QoS BronzeServiceClassDiffServ Sets the DiffServ value for Network service class content The valid range is 0 to 56 The default value is 48 Sets the DiffServ value for Premium Media service class content only if IPDiffServ is not set in the selected IP Profile The valid range is 0 to 56 The default value is 46 Note The value for the Premium Control DiffServ is determined by according to priority 1 IPDiffServ value in the selected IP Profile 2 PremiumServiceClassMediaDiffServ Sets the DiffServ value for Premium Control service class content only if ControllPDiffserv is not set in the selected IP Profile The valid range is 0 to 56 The default value is 46 Note The value for the Premium Control DiffServ is determined by according to priority 1 ControlPDiffserv value in the selected IP Profile 2 PremiumServiceClassControlDiffServ Sets the DiffServ value for the Gold service class content The valid range is 0 to 56 The default value is 26 Sets the DiffServ value for the Bronze service class content The valid range is 0 to 56 The default value is 10 SIP User s Manual 140 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 1 2 Configuring the Application Settings gt To configure the Application Settings parameters take these 4 steps 1 Op
199. aPack Table 5 41 Network Settings ini File Parameters continues on pages 149 to 151 ini File Parameter Name DisableNAT EnablelPAddrTranslation EnableUDPPortTranslation NoOperationSendingMode RTPNoOpEnable Valid Range and Description Enables disables the Network Address Translation NAT mechanism 0 Enabled 1 Disabled default Note The compare operation that is performed on the IP address is enabled by default and is controlled by the parameter EnablelPAddrTranslation The compare operation that is performed on the UDP port is disabled by default and is controlled by the parameter EnableUDPPortTranslation 0 Disable IP address translation 1 Enable IP address translation for RTP RTCP and T 38 packets default 2 Enable IP address translation for ThroughPacket 3 Enable IP address translation for all protocols RTP RTCP T38 and ThroughPacket When enabled the gateway 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 Consequently the remote IP address of the outgoing stream is replaced by the source IP address of the first incoming packet Note The NAT mechanism must be enabled for this parameter to take effect DisableNAT 0 0 Disable UDP port translation default 1 Enable UDP port translation When
200. accept an MWI NOTIFY message that indicates waiting messages or indicates that the MWI is cleared Supports 3 Way Conference using an external media server For more updated information on the gateway s supported features refer to the latest MP 11x amp MP 124 SIP Release Notes SIP User s Manual 24 Document LTRT 65408 SIP User s Manual 2 MediaPack Physical Description 2 MediaPack Physical Description This section provides detailed information on the hardware the location and functionality of the LEDs buttons and connectors on the front and rear panels of the MP 11x refer to Section 2 1 below and MP 124 refer to Section 2 2 on page 27 gateways For detailed information on installing the MediaPack refer to Chapter 0 on page 29 2 1 MP 11x Physical Description 2 1 1 MP 11x Front Panel Figure 2 1 illustrates the front layout of the MP 118 almost identical on MP 114 and MP 112 Table 2 1 lists and describes the front panel LEDs on the MP 11x Tip MP 11x FXS FXO gateways feature similar front panel LEDs they only differ in the number of Channels Status LEDs which correspond to the number of channels Figure 2 1 MP 118 Front Panel Connectors LLL LLL LLL LLL LLL LEAL LAL EAL ALLL LEE IE CT EE TNL LN LD ID LA a p A a en M118 VoIP Gatewsy 2 4 ay an ee a 3 Channels Status Version 5 0 25 December 2006 7a Ta wt AudioCodes MediaPack Table 2 1 Definition of MP 11x Front P
201. ach endpoint default Per gateway 1 Single Registration amp Authentication for the gateway Per Ch Select Mode 2 N A Usually Authentication on a per endpoint basis is used for FXS gateways in which each endpoint registers and authenticates separately with its own username and password Single Registration and Authentication Authentication Mode 1 is usually defined for FXO gateways Version 5 0 71 December 2006 7a Ta e wt AudioCodes MediaPack 5 5 1 3 Coders From the Coders screen you can configure the first to fifth preferred coders and their attributes for the gateway The first coder is the highest priority coder and is used by the gateway whenever possible If the far end gateway cannot use the coder assigned as the first coder the gateway attempts to use the next coder and so forth gt To configure the gateway s coders take these 9 steps 1 Open the Coders screen Protocol Management menu gt Protocol Definition submenu gt Coders option the Coders screen is displayed Figure 5 7 Coders Screen Coders Packetization Payload Silence Time Type Suppression 6 729 20 ME JIE Disabled gt Coder Name 6723 1 z 30 z 5 3 ip Disabled gt G 711A law 20 64 vif Disabled gt 2 From the Coder Name drop down list select the coder you want to use For the full list of available coders and their corresponding attributes refer to Table 5 5 3 Fr
202. active alarm table m The device has a mechanism to allow a manager to detect lost alarm raise and clear notifications sequence number in trap current sequence number MIB object m The device has a mechanism to allow a manager to recover lost alarm raise and clear notifications maintains a log history E The device sends a cold start trap to indicate that it is starting This allows the EMS to synchronize its view of the device s active alarms When the SNMP alarm traps are sent the carrier grade alarm system does not add or delete alarm traps as part of the feature This system provides the mechanism for viewing of history and current active alarm information 14 2 1 Active Alarm Table The device maintains an active alarm table to allow a manager to determine which alarms are currently active in the device Two views of the active alarm table are supported by the agent E acActiveAlarmTable in the enterprise acAlarm E alarmActiveTable and alarmActiveVariableTable in the IETF standard ALARM MIB rooted in the AC tree The acActiveAlarmTable is a simple one row per alarm table that is easy to view with a MIB browser The ALARM MIB is currently a draft standard and therefore has no OID assigned to it In the current software release the MIB is rooted in the experimental MIB subtree In a future release after the MIB has been ratified and an OID assigned it is to move to the official OID 14 2 2 Alarm History The device
203. ad type negotiation DTMF out of band transfer using e INFO method lt draft choudhuri sip info digit 00 txt gt e INFO method compatible with Cisco gateways e NOTIFY method lt draft mahy sipping signaled digits 01 txt gt Version 5 0 23 December 2006 a WH wt AudioCodes MediaPack SIP URL sip phone number IP address such as 1225556 10 1 2 4 where 122556 is the phone number of the source or destination or sip phone_number domain name such as 122556 myproxy com Note that the SIP URI host name can be configured differently per called number Supports RFC 4040 RTP payload format for a 64 kbit s transparent data Can negotiate coder from a list of given coders Supports negotiation of dynamic payload types Supports multiple ptime values per coder Supports RFC 3389 RTP Payload for Comfort Noise Supports RFC 3824 Using E 164 numbers with SIP ENUM Supports reception and DNS resolution of FQDNs received in SDP Supports lt draft ietf sip gruu O9 gt Obtaining and Using Globally Routable User Agent UA URIs GRUU in SIP Responds to OPTIONS messages both outside a SIP dialog and in mid call Generates SIP OPTIONS messages as Proxy keep alive mechanism Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS requests Implementation of MWI IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to the MWI server The MediaPack FXS gateways can
204. address you add must be your own terminal s IP address If it isn t further access from your terminal is denied Delete your terminal s IP address from the Web amp Telnet Access List last If it is deleted before the last access from your terminal is denied from the point of its deletion on Table 5 48 Web amp Telnet Access List ini File Parameter Parameter Name in ini File Parameter Format WebAccessList_x WebAccessList_0 10 13 2 66 WebAccessList_1 10 13 77 7 The default value is 0 0 0 0 the gateway can be accessed from any IP address Note This parameter can appear up to ten times SIP User s Manual 170 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 5 3 Configuring the Firewall Settings The MediaPack accommodates an internal Firewall allowing the security administrator to define network traffic filtering rules For detailed information on the internal Firewall refer to Section 12 5 on page 297 gt 1 Selected IsRule Source Local Packet Burst Rule Active IP To create a new access rule take these 6 steps Open the Firewall Settings screen Advanced Configuration menu gt Security Settings gt Firewall Settings option the Firewall Settings screen is displayed Figure 5 49 Firewall Settings Screen Action Upon Match Mask Protocol i Byte rate Port Range Size Bytes 0 mgmt customer com 255 255 255 255 0 80 tcp 0 0 0 ALLOW 1 192 0 0 0
205. ager and the NE refer to the same database to retrieve information or configure parameters This database is referred to as the Management Information Base MIB and is a set of statistical and control values Apart from the standard MIBs documented in IETF RFCs SNMP additionally enables the use of private MIBs containing a non standard information set specific functionality provided by the NE Directives issued by the SNMP Manager to an SNMP Agent consist of the identifiers of SNMP variables referred to as MIB object identifiers or MIB variables along with instructions to either get the value for that identifier or set the identifier to a new value configuration The SNMP Agent can also send unsolicited events towards the EM called SNMP traps The definitions of MIB variables supported by a particular agent are incorporated in descriptor files written in Abstract Syntax Notation ASN 1 format made available to EM client programs so that they can become aware of MIB variables and their use The device contains an embedded SNMP Agent supporting both general network MIBs such as the IP MIB VoP specific MIBs such as RTP and our proprietary MIBs acBoard acGateway acAlarm and other MIBs enabling a deeper probe into the inter working of the device All supported MIB files are supplied to customers as part of the release About SNMP SNMP Message Standard Four types of SNMP messages are defined Get A request that returns th
206. aintain this parameter s confidentiality On the Embedded Web Server a list of asterisks is displayed instead of the pre shared key On SNMP the pre shared key parameter is a write only parameter and cannot be read In the ini file the following measures to assure the secrecy of the IPSec and IKE tables are taken E Hidden IPSec and IKE tables When uploading the ini file from the gateway the IPSec and IKE tables are not available Instead the notifications shown in Figure 12 6 are displayed Figure 12 6 Example of an ini File Notification of Missing Tables TABLE IPSEC IKEDB_ TABLE This table contains hidden elements and will not be exposed This table exists on board and will be saved during restarts TABLE ITPSEC SPED TABLE table contains hidden elements and will not be exposed table exists on board and will be saved during restarts m Preserving the values of the parameters in the IPSec and IKE tables from one ini file loading to the next The values configured for the parameters in the IPSec tables in the ini file are preserved from one loading to another If a newly loaded ini file doesn t define IPSec tables the previously loaded tables remain valid To invalidate a previously loaded ini file s IPSec tables load a new ini file with an empty IPSec table shown below Figure 12 7 Empty IPSec IKE Tables IPSec_IKEDB_Table IPSec_IKEDB_ Table IPSEC_SPD_TABLE IPSEC_SPD_TABLE Ver
207. all Progress Tone File Name usa_tones_11 dat FXS Coefficient File Name MP11x10 1 fxs dat 11 Click the End Process button the Quick Setup screen appears and the full Web application is reactivated Version 5 0 201 December 2006 Ta WH wt AudioCodes MediaPack 5 9 2 Auxiliary Files The Auxiliary Files screen enables you to load to the gateway the following files Call Progress Tones coefficient Prerecorded Tones PRT and User Information The Voice Prompts file is currently not applicable to the MediaPack For detailed information on these files refer to Section 6 on page 209 For information on deleting these files from the MediaPack refer to Section 5 7 3 on page 193 Table 5 63 presents a brief description of each auxiliary file File Type Table 5 63 Auxiliary Files Descriptions Description FXS Coefficient This file contains the telephony interface configuration information for the VoIP gateway Call Progress Tones This information includes telephony interface characteristics such as DC and AC impedance feeding current and ringing voltage This file is specific to the type of telephony interface that the VoIP gateway supports In most cases you have to load this type of file Note Use the parameter CountryCoefficients described in Table 5 35 on page 132 to configure the FXO coefficients This is a region specific telephone exchange dependent file that contains the Call Progres
208. ame 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 AlarmHistoryTableMaxSize Determines the maximum number of rows in the Alarm History table The 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 SIP User s Manual 186 Document LTRT 65408 SIP User s Manual 5 Web Management 5 7 5 7 1 Status amp Diagnostics Use this menu to view and monitor the gateway s channels Syslog messages hardware software product information and to assess the gateway s statistics and IP connectivity information Gateway Statistics Use the screens under Gateway Statistics 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 Notes The Gateway Statistics screens doesn t refresh automatically To view updated information re access the screen you require 5 7 1 1 IP Connectivity The IP Connectivity screen provides you with an online read only network diagnostic connectivity information on all destination IP addresses configured in the Tel to IP Routing table This information is available only if the parameter AltRoutingTel2I
209. ameters configurable by the Embedded Web Server refer to Section 5 4 4 on page 54 Control Protocol the MediaPack control protocol e g SIP 5 4 1 Main Menu Bar The main menu bar of the Web Interface is divided into the following menus Quick Setup Use this menu to configure the gateway s basic settings for the full list of configurable parameters go directly to Protocol Management and Advanced Configuration menus An example of the Quick Setup configuration is described in Section 4 2 3 on page 43 m Protocol Management Use this menu to configure the gateway s control protocol parameters and tables refer to Section 5 5 on page 56 m Advanced Configuration Use this menu to set the gateway s advanced configuration parameters for advanced users only refer to Section 5 6 on page 137 E Status amp Diagnostics Use this menu to view and monitor the gateway s channels Syslog messages hardware software product information and to assess the gateway s statistics and IP connectivity information refer to Section 5 7 on page 187 Software Update Use this menu when you want to load new software or configuration files onto the gateway refer to Section 5 9 on page 197 Maintenance Use this menu to remotely lock unlock the device refer to Section 5 10 on page 204 save configuration changes to the non volatile flash memory refer to Section 5 10 2 on page 205 and remotely reset the gateway
210. an MediaPack FXO gateway This ini file can then be converted to a dat file that can be loaded to the gateway using the TrunkPack Downloadable Conversion utility To use this wizard a MediaPack FXO gateway connected to your PBX with two physical phone lines is required This gateway must be configured with factory default settings and shouldn t be used for phone calls during the operation of the wizard Note that firmware version 4 2 and above is required on the gateway Version 5 0 367 December 2006 7a Ta P wt AudioCodes MediaPack E 2 2 Installation The CPTWizard can be installed on any Windows 2000 or Windows XP based PC Windows compliant networking and audio peripherals are required for full functionality To install the CPTWizard copy the files from the supplied installation kit to any folder on your PC No further setup is required approximately 5 MB of hard disk space are required E 2 3 Initial Settings gt To start the CPTWizard take these 5 steps 1 Execute the CPTWizard exe file the wizard s initial settings screen is displayed Figure E 6 Initial Settings Screen fe AudioCodes Call Progress Tones Wizard Welcome to the AudioCodes Call Progress Tones Wizard 7a i AudioCodes Please enter the IP address of an MP 10x FXO Gateway f10 31 4 49 Select two active ports to be used and enter their phone numbers below Port fi Phone Number 2001 Port 2 Phone Number 2002 Invalid phon
211. an appear up to eight times for 8 port gateways and up to 24 times for MP 124 Version 5 0 123 December 2006 Ta A wi AudioCodes MediaPack 5 5 9 5 Call Forward The VoIP gateway allows you to forward incoming IP gt Tel calls using 302 response based on the VoIP gateway port to which the call is routed applicable only to FXS gateways The Call Forwarding Table is applicable only if the Call Forward feature is enabled To enable Call Forward set Enable Call Forward to Enable in the Supplementary Services screen or EnableForward 1 in the ini file refer to Table 5 9 gt To configure the Call Forward table take these 4 steps 1 Open the Call Forward Table screen Protocol Management menu gt Endpoint Settings submenu gt Call Forward option the Call Forward Table screen is displayed Figure 5 31 Call Forward Table Screen Call Forward Table Gateway Forward Forward to Time for No Port Type Phone Number Reply Forward Eom 1 FXS On busy v 201 30 Port 2 FXS On busy v 201 30 Port 3 FXS No Answer 203 30 Port 4 FXS Unconditional 202 10 2 1 1f 30 Pot 5 FXO Deactivate 30 Pot 6 FXO Deactivate 30 Port FXO Deactivate 30 Port 8 Deactivate 30 2 Configure the Call Forward parameters for each port acc
212. anel LED Indicators continues on pages 26 to 26 LED Type Color State Definition Channels Telephone Blinking The phone is ringing incoming call before answering Status Interface Fast Blinking Line malfunction Off Normal onhook position On Offhook Ringing Uplink Ethernet On Valid 10 100 Base TX Ethernet connection Link Status Off No uplink Fail Failure On Failure fatal error Indication Or system initialization Off Normal working condition Ready Device On Device powered self test OK ae Off Software loading or System failure Power Power On Power iscurrently being supplied to the device Suppl PAA Either there s a failure disruption in the AC power supply Off or power is currently not being supplied to the device through the AC power supply entry 2 1 2 MP 11x Rear Panel Figure 2 2 illustrates the rear layout of the MP 118 almost identical on MP 114 and MP 112 Table 2 2 lists and describes the rear panel connectors and button on the MP 11x Figure 2 2 MP 118 Rear Panel Connectors 100 240 0 3A max 50 60Hz Table 2 2 MP 11x Rear Panel Component Descriptions Item Label Component Description 100 240 0 3A max AC power supply socket 2 Ethernet 10 100 Base TX Uplink port 3 RS 232 RS 232 status port requires a DB 9 to PS 2 adaptor 4 FXS or FXO 2 4 or 8 FXS FXO ports 5 Reset Reset button SIP User s Manual 26 Document LTRT 65408 SIP User s Manual 2 MediaPack Physical Descri
213. ant Proxy IP Address ProxylIP Third Redundant Proxy IP Address ProxylIP Description IP addresses of the first redundant Proxy you are using Enter the IP address as FQDN or in dotted format notation for example 192 10 1 255 You can also specify the selected port in the format lt IP Address gt lt port gt Note 1 This parameter is available only if you select Use Proxy in the Enable Proxy field Note 2 When port number is specified DNS NAPTR SRV queries aren t performed even if ProxyDNSQueryType is set to 1 ini file note The IP address of the first redundant Proxy is defined by the second repetition of the ini file parameter ProxylP IP addresses of the second redundant Proxy you are using Enter the IP address as FQDN or in dotted format notation for example 192 10 1 255 You can also specify the selected port in the format lt IP Address gt lt port gt Note 1 This parameter is available only if you select Use Proxy in the Enable Proxy field Note 2 When port number is specified DNS NAPTR SRV queries aren t performed even if ProxyDNSQueryType is set to 1 ini file note The IP address of the second redundant Proxy is defined by the third repetition of the ini file parameter ProxylP IP addresses of the third redundant Proxy you are using Enter the IP address as FQDN or in dotted format notation for example 192 10 1 255 You can also specify the s
214. anual 196 Document LTRT 65408 SIP User s Manual 5 Web Management 5 9 Software Update The Software Update menu enables users to upgrade the MediaPack software by loading a new cmp file along with the ini and a suite of auxiliary files or to update the existing auxiliary files The Software Update menu comprises two submenus E Software Upgrade Wizard refer to Section 5 9 1 below E Load Auxiliary Files refer to Section 5 9 2 on page 202 Note When upgrading the MediaPack software you must load the new cmp file with all other related configuration files 5 9 1 Software Upgrade Wizard The Software Upgrade Wizard guides users through the process of software upgrade selecting files and loading them to the gateway The wizard also enables users to upgrade software while maintaining the existing configuration Using the wizard obligates users to load and burn a cmp file Users can choose to also use the Wizard to load the ini and auxiliary files e g Call Progress Tones but this option cannot be pursued without loading the cmp file For the ini and each auxiliary file type users can choose to reload an existing file load a new file or not load a file at all Warning 1 The Software Upgrade Wizard requires the MediaPack to be reset at the end of the process disrupting any of its traffic To avoid disruption disable all traffic on the MediaPack before initiating the Wizard Warning 2 Verify prior to click
215. ards the FXO gateway The FXO gateway detects these pulses and relays them over IP to the FXS gateway using a proprietary INFO messages shown in Figure 11 2 The FXS gateway generates the same pulses to the connected phone The parameter MeteringType described in Table 5 35 is used to determine the frequency of the metering tone 12 kHz default or 16 kHz In addition the correct 12 or 16 kHz coefficient must be used for both FXS and FXO gateways To enable this feature configure SendMetering2IP 1 The proprietary INFO message used to relay the metering tone pulse contains a Content Type message Metering Figure 11 2 Proprietary INFO Message for Relaying Metering Tones INNO SijoslOSeio 13 Ato STE 2 0 Via SIP 2 0 UDP 10 13 83 2 branch z9hG4bKacEizRjAa Max Forwards 70 etom Wenyaterell lt sip ZOE iS tS 2 peer Leuk Sieve TS To lt sip 108 10 13 83 1 user phone gt tag 1c1412617336 CalLI 1Dg ZOSLOLMIZ CCC 13 c3 2 CSeq 3 INFO Conisalcirm lt st pre OMG OP om SS Supported em timer replaces path Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO SUBSCRIBE UPDATE User Agent Audiocodes Sip Gateway MP 114 FXS v 4 40 0 18700 Content Type message Metering Content Length 0 Version 5 0 217 December 2006 7 T E MediaPack Reader s Notes SIP User s Manual 278 Document LT RT 65408 SIP User s Manual 12 Security 12
216. are used in the Tel and IP Profile Settings screens to assign different coders to Profiles For each group you can define the first to fifth preferred coders and their attributes for the gateway The first coder is the highest priority coder and is used by the gateway whenever possible If the far end gateway cannot use the coder assigned as the first coder the gateway attempts to use the next coder and so forth SIP User s Manual 108 Document LTRT 65408 SIP User s Manual 5 Web Management 10 11 To configure the coder group settings take these 11 steps Open the Coder Group Settings screen Protocol Management menu gt Profile Definitions submenu gt Coder Group Settings option the Coder Group Settings screen is displayed Figure 5 22 Coder Group Settings Screen Coder Group Settings Coder Group ID Packetization Payload Silence Time Type Suppression G71Alew Ji wea Disabled z 30 v 53 i Disabled z Coder Name From the Coder Group ID drop down list select the coder group you want to edit up to four coder groups can be configured From the Coder Name drop down list select the coder you want to use For the full list of available coders and their corresponding attributes refer to Table 5 5 From the Packetization Time drop down list select the packetization time in msec for the coder you selected The packetization time determines how many coder payloads
217. arning Ensure that you connect FXS ports to analog telephone or to PBX trunk lines only and FXO ports to CO PBX lines only The MediaPack is supplied as a sealed unit and must only be serviced by qualified service personnel Warning Disconnect the MediaPack from the mains and from the Telephone Network Voltage TNV before servicing gt gt gt FP PP SIP User s Manual 18 Document LTRT 65408 SIP User s Manual 1 Overview 1 Overview This document provides you with the information on installation configuration and operation of the VoIP analog gateways listed in the table below Table 1 1 Supported Product Configurations Product Name FXS ce Se E one MP 124 v x i MP 118 v fi es 2 MP 114 4 Ete i MP 112 v a 4 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 As these units have similar functionality with the exception of their number of channels and some minor features they are collectively referred to throughout this manual as the MediaPack 1 1 Gateway Description The MediaPack series analog VolP gateways are cost effective cutting edge technology products These stand alone analog VolP gateways provide superior voice technology for connecting legacy telephones fax machines and PBX systems with IP based telephony networks as well as for integration with new IP based PBX architecture These products
218. associate an alternative IP address to called telephone number prefix assign it with an additional entry with a different IP address or use an FQDN that resolves to two IP addresses Call is sent to the alternative destination when one of the following occurs e No ping to the initial destination is available or when poor QoS delay or packet loss calculated according to previous calls is detected or when a DNS host name is not resolved For detailed information on Alternative Routing refer to Section 8 7 on page 231 e When a release reason that is defined in the Reasons for Alternative Tel to IP Routing table is received For detailed information on the Reasons for Alternative Routing Tables refer to Section 5 5 5 5 on page 106 Alternative routing using this table is commonly implemented when there is no response to an INVITE message after INVITE retransmissions The gateway then issues an internal 408 No Response implicit release reason If this reason is included in the Reasons for Alternative Routing table the gateway immediately initiates a call to the redundant destination using the next matched entry in the Tel to IP Routing table Note that if a domain name in this table is resolved to two IP addresses the timeout for INVITE retransmissions can be reduced by using the parameter Number of RTX Before Hotswap SIP User s Manual 100 Document LTRT 65408 SIP User s Manual 5 Web Manage
219. ately dials the destination telephone number In other words the IP caller doesn t dial the PSTN number upon hearing a dial tone Figure 8 1 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 Is VaitForDialTone and VYVaitForDialTone F4 200 OK immediatley or after detecting polarity reversal or voice Version 5 0 223 December 2006 Ta WH wi AudioCodes MediaPack One stage dialing incorporates the following FXO functionality m Waiting for Dial Tone The Waiting for Dial Tone feature enables the gateway 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 E Time to Wait Before Dialing The Time to Wait Before Waiting feature 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 Answer Supervision The Answer Supervision feature enables the FXO gateway to determine when a call is connected by using one of the following methods e Polarity Reversal the gateway sends a 200 OK in response to an INVITE only when it detects a polarity reversal e Voice Detection the gateway sends a 200 OK in response to an INVITE only when
220. atile memory and used when BootP is inaccessible BootP can be used again to change the IP address of the MediaPack for example BootP DHCP Server Parameters BootP DHCP can be used to provision the following parameters included in the BootP DHOP reply Note that only the IP address and subnet mask are mandatory m P address subnet mask These mandatory parameters are sent to the MediaPack every time a BootP DHCP process occurs E Default gateway IP address An optional parameter that is sent to the MediaPack only if configured in the BootP DHCP server E TFTP server IP address An optional parameter that contains the address of the TFTP server from which the firmware cmp and ini files are loaded m DNS server IP address primary and secondary Optional parameters that contain the IP addresses of the primary and secondary DNS servers These parameters are available only in DHCP and from Boot version 1 92 E Syslog server IP address An optional parameter that is sent to the MediaPack only if configured This parameter is available only in DHCP E SIP server IP address Two optional parameters that are sent to the MediaPack only if configured These parameters are available only in DHCP m Firmware file name An optional parameter that contains the name of the firmware file to be loaded to the gateway via TFTP E jnifile name An optional parameter that contains the name of the ini file to be loaded to the gateway
221. ating support for V 152 v 0 o 0 0 IN IPV4 lt IPAdressA gt ges t 0 0 p 1 c IN IP4 lt IPAddressA m audio lt udpPort A gt RTP AVP 18 0 a ptime 10 a rtpmap 96 PCMU 8000 a gpmd 96 vbd yes In the example above V 152 implementation is supported using the dynamic payload type 96 and G 711 mu law as the VBD codec as well as the voice codecs G 711 mu 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 SIP User s Manual 222 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 4 8 4 1 8 4 1 1 FXO Operating Modes This section provides a description of the FXO operating modes and gateway configurations for Tel to IP and IP to Tel calls IP to Telephone Calls The FXO gateway provides the following FXO operating modes for IP to Tel calls mE One stage dialing e Waiting for dial tone e Time to wait before dialing e Answer supervision m Two stage dialing BH Dialing time e Disconnect supervision e DID wink One Stage Dialing One stage dialing is when the FXO gateway receives an IP to Tel call off hooks the PBX line connected to the telephone and then immedi
222. ation 3 Enable access to the following directories and files by typing e gt chmod 777 perl e gt chmod 755 put cgi e gt chmod 777 html the name of the server s shared files directory 4 Configure the Apache sever a Open etc httpd conf httpd conf or a similar file for editing b Set the KeepAlive parameter to true SIP User s Manual 384 Document LTRT 65408 SIP User s Manual G Installation and Configuration of Apache HTTP Server c Set the MaxKeepAliveRequests parameter to 0 enables an unlimited number of requests during a persistent connection required for multiple consecutive HTTP POST requests for uploading the file d Set MaxClients to 250 e Change the mod_perl module lines to lt IfModule mod perl c gt Alias perl var www perl lt Directory var www perl gt SetHandler perl script PerlHandler Apache Registry Options ExecCGI PerlSendHeader On lt Directory gt lt IfModule gt Script PUT perl put cgi Version 5 0 385 December 2006 7 T E Reader s Notes SIP User s Manual 386 Document LTRT 65408 SIP User s Manual H Regulatory Information H Regulatory Information Declaration of Conformity Application of Council Directives 73 23 EEC including amendments 89 336 EEC including amendments 1999 5 EC Annex ll of the Directive Standards to which Conformity is Declared EN55022 1998 A1 2000 A2 2003 EN55024 1998 A1 2001 A2
223. ation to correspond with your network IP settings If your network doesn t feature a default gateway enter a dummy value in the Default Gateway IP Address field Click the Reset button and then at the prompt click OK the MediaPack applies the changes and restarts Record and retain the IP address and subnet mask you assign the MediaPack Do the same when defining new username or password If the Embedded Web Server is unavailable for example if you ve lost your username and password use the BootP TFTP Trivial File Transfer Protocol configuration utility to access the device reflash the load and reset the password refer to Appendix C on page 349 for detailed information on using a BootP TFTP configuration utility to access the device Disconnect your PC from the MediaPack or from the hub switch depending on the connection method you used in Step 1 Reconnect the MediaPack and your PC if necessary to the LAN Restore your PC s IP address amp subnet mask to what they originally were If necessary restart your PC and re access the MediaPack via the Embedded Web Server with its new assigned IP address 4 2 2 Assigning an IP Address Using BootP BootP procedure can also be performed using any standard compatible BootP server You can also use BootP to load the auxiliary files to the MediaPack refer to Section 5 9 2 1 on page 203 SIP User s Manual 42 Document LTRT 65408 SIP User s Ma
224. ation number prefix h Calling number presentation 0 to allow presentation 1 to restrict presentation The b to f and h manipulation rules are applied if the called and calling numbers match the a and g conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example SourceNumberMapTel2IP 01 2 972 0 0 1 SourceNumberMapTel2IP 03 2 667 0 0 22 Note 1 Presentation is set to Restricted only if Asserted Identity Mode is set to P Asserted Note 2 Number Plan amp Type can optionally be used in Remote Party ID RPID header by using the EnableRPlHeader parameter SIP User s Manual 94 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 14 Number Manipulation ini File Parameters continues on pages 93 to 95 Parameter Description SourceNumberMapIP2Tel Manipulate the source number for IP to Tel calls 5 5 3 1 NumberMapIP2Tel a b c d e f g h i a Source number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Notin use should be set to f Notin use should be set to g Destinatio
225. ault Yes 1 FXO gateways send a metering tone INFO message to IP on detection of 12 16 kHz metering pulse FXS gateways generate the 12 16 kHz metering tone on reception of a metering message Note 1 Suitable 12 kHz or 16 kHz coeff must be used for both FXS and FXO gateways The MeteringType parameter must be defined in both FXS FXO gateways Note 2 The proprietary metering tone INFO message is shown in Section 11 on page 277 No 0 Call isn t released FXO gateway Yes 1 Call is released on FXO gateways if busy or reorder fast busy tones are detected on the gateway s FXO port default Defines the time interval in seconds after a call has ended and a new call can be accepted for IP to Tel calls Applicable only to FXO gateways The valid range is 0 to 10 The default value is 1 second Note Occasionally after a call is ended and onhook is applied a delay is required before placing a new call and performing offhook This is necessary to prevent wrong hook flash detection or other glare phenomena FXO gateways can disconnect a call after a dial tone from the PBX is detected No 0 Call isn t released Yes 1 Call is released if dial tone is detected on the gateway s FXO port default Note This option is in addition to the mechanism that disconnects a call when either busy or reorder tones are detected Version 5 0 129 December 2006 Ta A wi AudioCodes MediaPack 5
226. b Server itself the access level determines the extent of the access i e availability of screens and read write privileges Note that additional accounts can be defined using a RADIUS server refer to Section 12 3 on page 293 Version 5 0 49 December 2006 Ta WH wt AudioCodes MediaPack Table 5 1 lists the available access levels and their privileges Table 5 1 Available Access Levels and their Privileges Numeric Access Level Representation Privileges Security Administrator 200 Read write privileges for all screens Administrator 100 Read only privilege for security related screens and read write privileges for the others User Monitor 50 No access to security related and file loading screens and read only access to the others No Access 0 No access to any screen 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 access level mechanism operation is as follows for both Web and RADIUS accounts Each Web screen features two hard coded minimum access levels read and write The read access level determines whether the screen can be viewed The write access level determines whether the information in the screen can be modified When a user tries to access a specific Web screen the user s access level is compared with the access levels of the screen E fthe access level of the user is less than the screen s read acce
227. before releasing TimeForReorderTone Answer Supervision EnableVoiceDetection Rings before Detecting Caller ID RingsBeforeCallerID the line The valid range is 0 to 100 The default is 0 seconds Usually after playing a Reorder Busy tone for the specified duration the FXS gateway starts playing an Offhook Warning tone Note 1 Selection of Busy or Reorder tone is performed according to the release cause received from IP Note 2 Refer also to the parameter CutThrough described in Table 5 8 Yes 1 FXO gateway sends 200 OK to INVITE messages when speech fax modem is detected No 0 200 OK is sent immediately after the FXO gateway finishes dialing default Usually this feature is used only with early media establish voice path before the call is answered Note This feature is applicable only to One Stage dialing Sets the number of rings before the gateway starts detection of Caller ID FXO only 0 0 Before first ring 1 1 After first ring default 2 2 After second ring SIP User s Manual 128 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 33 FXO Parameters continues on pages 127 to 130 Parameter Send Metering Message to IP SendMetering2IP Disconnect on Busy Tone DisconnectOnBusyTone Guard Time Between Calls GuardTimeBetweenCalls Disconnect on Dial Tone DisconnectOnDialTone Description No 0 Disabled def
228. ble Always Use Proxy Disable Send All Invite to Proxy No Enable Proxy Hot Swap Disable Enable Registration Disable Gateway Name audiocodes com Gateway Registration Name DNS Query Type A Record Proxy DNS Query Type A Record Subscription Mode Per Endpoint Use Gateway Name for OPTIONS No Number of RTX Before Hot Swap 3 User Name Password Cnonce 0a123bcf Authentication Mode Per Endpoint Version 5 0 65 December 2006 Ta WH wt AudioCodes MediaPack 2 Configure the Proxy amp Registration parameters according to Table 5 4 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 so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 4 Proxy amp Registration Parameters continues on pages 66 to 71 Parameter Enable Proxy IsProxyUsed Proxy Name ProxyName Proxy IP Address ProxyIP Gateway Name SIPGatewayName Gateway Registration Name GWRegistrationName Description Don t Use Proxy 0 Proxy isn t used the internal routing table is used instead default Use Proxy 1 Proxy is used If you are using a Proxy server enter the IP a
229. ble 3 4 For detailed information on the MP 124 rear panel connectors refer to Section 2 2 2 on page 28 Table 3 4 MP 124 Cables and Cabling Procedure Cable Cabling Procedure Protective Connect an earthed strap to the chassis protective earthing screw 6 32 UNC screw and earthing strap fasten it securely according to the safety standards Connect the Ethernet connection on the MP 124 directly to the network using a crossover RJ 45 Ethernet RJ 45 Ethernet cable For connector pinouts refer to Figure 3 11 below cable Note that when assigning an IP address to the MP 124 using HTTP under Step 1 in Section 4 2 1 you may be required to disconnect this cable and re cable it differently Refer to the MP 124 Safety Notice below Wire the 50 pin Telco connectors according to the pinouts in Figure 3 12 on page 38 50 pin Telco and Figure 3 13 on page 39 cable MP 124 Attach each pair of wires from a 25 pair Octopus cable to its corresponding socket on devices only the MDF Adaptor Block s rear Connect the wire pairs at the other end of the cable to a male 50 pin Telco connector An Octopus Insert and fasten this connector to the female 50 pin Telco connector on the MP 124 cable is not rear panel labeled Analog FXS Lines 1 24 included with the Connect the telephone lines from the Adaptor Block to a fax machine modem or MP 124 package telephones by inserting each RJ 11 connector on the 2 wire line cords of the POTS phones into
230. box to the right of that parameter This clears the parameter provided by the template and enables you to edit the entry Clicking the check box again restores the template settings 5 Click Apply to save this entry to the list of clients or click Apply amp Reset to save this entry to the list of clients and send a reset message to that gateway to immediately implement the settings Note To use Apply amp Reset you must enable ARP Manipulation in the Preferences window Also you must have administrator privileges for the computer you are using C 11 2 Deleting Clients gt To delete a client from the BootP Tool take these 3 steps 1 Select the client that you wish to delete by clicking on the line in the window for that client 2 Click the Delete Current Client button Ba a warning pops up 3 To delete the client click Yes C 11 3 Editing Client Parameters gt To edit the parameters for an existing client take these 4 steps 1 Select the client that you wish to edit by clicking on the line in the window for that client 2 Parameters for that client display in the parameter fields on the right side of the window 3 Make the changes required for each parameter 4 Click Apply to save the changes or click Apply amp Reset to save the changes and send a reset message to that gateway to immediately implement the settings Note To use Apply amp Reset you must enable ARP Manipulation in the Preferences window Also
231. bute VSA Value 1 Number Name No Ee Format anole AAA Request Attributes String Account number or calling up to 15 Start Acc d user Name party number or blank digits 9421383 VA Stop Acc long 4 NAS IP IP address of the requesting Numenc 192 168 14 Start Acc Address MediaPack 43 Stop Acc aa Start Acc 6 Service Type Type of service requested Numeric 1 login Stop Acc h323 3 Up to 32 Start Acc 26 incoming 1 H 323 SIP call identifier octets Stop Acc conf id 26 h323 remote 23 IP address of the remote N r ric Stop Acc address gateway 26 h323 conf id 24 H 323 SIP call identifier Up to 32 Start Acc 1 The values in column AAA are as follows Start Acc Start Accounting Stop Acc Stop Accounting Version 5 0 233 December 2006 r7 a T wt AudioCodes MediaPack Table 8 2 Supported RADIUS Attributes continues on pages 233 to 235 Attribute Attribute VSA Value 1 Number Name No Purpose Format eampr AAA octets Stop Acc 26 h323 setup 25 Setup time in NTP format 1 String Start Acc time Stop Acc The call s originator Answer 26 si 26 Answering IP or Originator String Originate eal foe 9 PSTN etc P mt Protocol type or family used Start Acc 26 h323 call type 27 on this leg of the call String VolP Stop Acc 26 h323 28 Connect time in NTP format String Stop Acc connect time h323 roe 26 disc nn t 29 Disconnect time in NTP String Stop Acc format time h323 26 disconnect 30 SA disconnect cau
232. cable only to MediaPack FXS gateways Determines the type of Direct Inward Dialing DID signaling support for Japan NTT modem DTMF or Frequency Shift Keying FSK based signaling Gateways 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 FSkK based signaling default 1 DTMF based signaling Note Applicable only to FXS gateways Threshold of the packet count in percents below which is considered silence by the media gateway The valid range is 1 to 100 The default is 8 Note Applicable only if silence is detected according to packet count FarEndDisconnectSilenceMethod 1 Defines that the T 38 packets are sent received using the same port as RTP 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 SIP User s Manual 132 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 35 Protocol Management ini File Parameters continues on pages 132 to 137 ini File Parameter Name DisableAutoDTMFMute FirstCallWaitingTonelD CounitryCoefficients EnableCallerIDTypeTwo MeteringType PolarityReversalType Valid Range and Description Enables disables the automatic mute of DTMF digits when out of band DTMF transmission is used 0 Auto mute is used default 1 No automatic mute of in band DTMF When
233. cations link with the MediaPack refer to Section 10 2 on page 262 Note 3 To configure the Syslog logging levels use the parameter Debug Level SNMP Seitings For detailed information on the SNMP parameters that can only be configured via the ini file refer to Table 5 57 on page 186 For detailed information on developing an SNMP based program to manage your devices refer to Chapter 14 on page 305 SNMP Managers Table Refer to Section 5 6 6 1 on page 178 SNMP Community Strings Refer to Section 5 6 6 2 on page 180 SNMP V3 Table Refer to Section 5 6 6 3 on page 181 Enable SNMP Enable 0 SNMP is enabled default DisableSNMP Disable 1 SNMP is disabled and no traps are sent Trap Manager Host Name Defines a FQDN of a remote host that is used as an SNMP Manager The SNMPTrapManagerHostName resolved IP address replaces the last entry in the trap manager table defined by the parameter SNMPManagerTablelP_x and the last trap manager entry of snmpTargetAddrTable in the snmpTargetMIB For example mngr corp mycompany com The valid range is a 99 character string Activity Types to Report via Activity Log Messages The Activity Log mechanism enables the MediaPack to send log messages to a Syslog server that report certain types of Web actions according to a pre defined filter The following filters are available Parameters Value Change ActivityListToLog PVC Auxiliary Files Loading ActivityList
234. ce Type Login User ACL Auth Level ACL Auth UserLevel 4 Record and retain the IP address port number shared secret vendor ID and VSA access level identifier if access levels are used used by the RADIUS server 5 Configure the gateway s relevant parameters according to Section 12 4 2 below 12 4 2 Configuring RADIUS Support For information on the RADIUS parameters refer to Table 5 50 on page 174 gt To configure RADIUS support on the gateway via the Embedded Web Server take these 12 steps 1 Access the Embedded Web Server refer to Section 5 3 on page 51 2 Open the General Security Settings screen Advanced Configuration menu gt Security Settings gt General Security Settings option the General Security Settings screen is displayed 3 Under section General RADIUS Settings in the field Enable RADIUS Access Control select Enable the RADIUS application is enabled 4 In the field Use RADIUS for Web Telnet Login select Enable RADIUS authentication is enabled for Web and Telnet login 5 Enter the RADIUS server IP address port number and shared secret in the relevant fields Version 5 0 295 December 2006 Ta WH wt AudioCodes MediaPack 10 11 12 Under section RADIUS Authentication Settings in the field Device Behavior Upon RADIUS Timeout select the gateway s operation if a response isn t received from the RADIUS ser
235. cess begins immediately The Current Admin State displays the current state LOCKED or UNLOCKED gt To unlock the gateway take these 2 steps 1 Access the Maintenance Actions screen as described above in the previous procedure 2 Click the UNLOCK button Unlock starts immediately and the gateway is ready for new incoming calls 5 10 2 Saving Configuration The Maintenance Actions screen enables you to save the current parameter configuration and the loaded auxiliary files to the non volatile memory i e flash so they are available after a hardware reset or power fail Parameters that are only saved to the volatile memory RAM revert to their previous settings after hardware reset Saving changes to the non volatile memory may disrupt traffic on the gateway To avoid this disable all new traffic before saving by performing a graceful lock refer to Section 5 10 1 on page 204 In the Web interface parameters prefixed with an exclamation mark are saved to the non volatile memory only after a device reset gt To save the changes to the non volatile take these 2 steps 1 Open the Maintenance Actions screen Maintenance menu the Maintenance Actions screen is displayed Figure 5 72 Maintenance Actions Screen Maintenance Actions Graceful Option Current Admin State Save Configuration Save Configuration 2 Click the BURN button a confirmation message
236. ck screws not supplied to attach the shelf to the rack SIP User s Manual 32 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack 3 1 5 Cabling the MP 11x Cable your MP 11x according to each section of Table 3 3 For detailed information on the MP 11x rear panel connectors refer to Table 2 2 on page 26 Table 3 3 MP 11x Cables and Cabling Procedure Cable Cabling Procedure RJ 45 Ethernet Connect the Ethernet connection on the MP 11x directly to the network using a crossover cable RJ 45 Ethernet cable For connector pinouts refer to Figure 3 4 below Note that when assigning an IP address to the MP 11x using HTTP under Step 1 in Section 4 2 1 you may be required to disconnect this cable and re cable it differently RJ 11 two wire Connect the RJ 11 FXS connectors to fax Ensure that FXS and FXO ports are telephone cords machines modems or phones connected to the correct devices otherwise damage can occur Connect the RJ 11 FXO connectors to telephone The RJ 11 connector pinouts is exchange analog lines or PBX extensions described in Figure 3 5 Lifeline For detailed information on setting up the Lifeline refer to the procedure in Section 3 1 5 2 on page 34 RS 232 serial cable For detailed information on connecting the MP 11x RS 232 port to your PC refer to Section 3 1 5 1 AC Power cable Connect the MP 11x power socket to the mains Warning To reduce the risk of fire use only No 26 AWG or large
237. conds The range is 10 to 7200 Duration in msec of the played Stutter dial tone which indicates that Call Forwarding is enabled or that there is a waiting message s The default is 2 000 i e 2 seconds The range is 1 000 to 60 000 The Stutter tone is played instead of a regular Dial tone when a Call Forward is enabled on the specific port or when MWI is received The tone is composed of a Confirmation tone which is played for a user defined duration StutterToneDuration followed by a Stutter tone Both tones are defined in the CPT file Note 1 This parameter is applicable only to FXS gateways Note 2 The message waiting notification MWI tone takes precedence over the call forwarding reminder tone For detailed information on Message Waiting Indication MWI refer to Section 8 1 6 on page 218 Enables or disables the 3 Way Conference feature Valid options include Disable 0 Disables 3 way conference feature default Enable 1 Enables 3 way conference feature Defines the digit pattern that once detected generates the Conference initiating INVITE when 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 gateway uses this identifier in the Conference initiating INVITE that is sent to the media server when Enable3WayConferenceis set to 1 The default value is conf
238. connect the call but instead signals to the gateway that the call is disconnected using one of the following methods Detection of polarity reversal current disconnect This is the recommended method The call is immediately disconnected after polarity reversal or current disconnect is detected on the Tel side assuming the PBX CO produces this signal Relevant parameters EnableReversalPolarity EnableCurrentDisconnect CurrentDisconnectDuration CurrentDisconnectDefaultThreshold and TimeToSampleAnalogLineVoltage Detection of Reorder Busy Dial tones The call is immediately disconnected after Reorder Busy Dial 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 unknown define them in the CPT file the tone produced by the PBX CO must be recorded and its frequencies analyzed refer to Section E 2 7 on page 374 This method is slightly less reliable than the previous one You can use the CPTWizard described in Section E 1 3 on page 366 to analyze Call Progress Tones generated by any PBX or telephone network Relevant parameters DisconnectOnBusyTone and DisconnectOnDialTone 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 t
239. connection is considered a failure The range is 1 to 20 The default value is 20 Transmission delay in msec at which the IP connection is considered a failure The range is 100 to 1000 The default value is 250 msec Version 5 0 99 December 2006 Ta WH wt AudioCodes MediaPack 5 5 5 2 Tel to IP Routing Table The Tel to IP Routing Table is used to route incoming Tel calls to IP addresses This routing table associates a called calling telephone number s prefixes with a destination IP address or with an FQDN Fully Qualified Domain Name When a call is routed through the VoIP gateway Proxy isn t used the called and calling numbers are compared to the list of prefixes on the IP Routing Table up to 50 prefixes can be configured Calls that match these prefixes are sent to the corresponding IP address If the number dialed does not match these prefixes the call is not made When using a Proxy server you do not need to configure the Tel to IP Routing Table However if you want to use fallback routing when communication with Proxy servers is lost or to use the Filter Calls to IP and IP Security features or to obtain different SIP URI host names per called number or to assign IP profiles you need to configure the IP Routing Table Note that for the Tel to IP Routing table to take precedence over a Proxy for routing calls set the parameter PreferRouteTable to 1 The gateway checks the D
240. d msec 400 Enable Early Media Disable Progress Indicator to IP No PI v Coder Group Default Coder Group 2 From the Profile ID drop down list select the Tel Profile you want to edit up to four Tel Profiles can be configured 3 In the Profile Name field enter a name that enables you to identify the Profile intuitively and easily Version 5 0 111 December 2006 Ta WH wt AudioCodes MediaPack Parameter TelProfile_ID From the Profile Preference drop down list select the preference 1 20 of the current Profile The preference option is used to determine the priority of the Profile Where 20 is the highest preference value 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_ID 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 an intersection of the coders is performed i e only common coders remain 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 the description of the screen in which it is configured as an indi
241. d that the columns User Name and User Password have Read Create permissions The column Time Connected has a Read permission and the column Permissions has Read Create Maintenance Write permissions Dynamic Tables vs Static Tables Static Tables Static tables don t support adding new lines or removing deleting existing lines All lines in a Static table are pre configured with default values Users can only modify the values of the existing lines After reset all lines in a Static table are available Dynamic Tables Dynamic tables support adding and removing lines They are always initialized as empty tables with no lines Users should add lines to a Dynamic table via the ini file or at run time via the Embedded Web Server for example Note Certain dynamic tables may initialize a line or more at start up If so it is explained in the specific table s documentation Secret Tables A table is defined as a secret table if it contains at least a single secret data field or if it depends on another secret table A secret data field is a field that mustn t be revealed to the user For example in the IPSec application IPSec tables are defined as secret tables as the IKE table contains a pre shared key that must be concealed Therefore the SPD table that depends on the IKE table is defined as a secret table as well There are two major differences between tables and secret tables m The secret field i
242. ddress of the primary Proxy server in the Proxy IP address field If you are not using a Proxy server you must configure the Tel to IP Routing table on the gateway described in Section 5 5 5 2 on page 100 Defines the Home Proxy Domain Name If specified the Proxy Name is used as Request URI in REGISTER INVITE and other SIP messages If not specified the Proxy IP address is used instead IP address and optionally port number of the primary Proxy server you are using Enter the IP address as FQDN or in dotted format notation for example 201 10 8 1 You can also specify the selected port in the format lt IP Address gt lt port gt This parameter is applicable only if you select Yes in the Is Proxy Used field If you enable Proxy Redundancy by setting EnableProxyKeepAlive 1 or 2 the gateway can work with up to four Proxy servers If there is no response from the primary Proxy the gateway tries to communicate with the redundant Proxies When a redundant Proxy is found the gateway either continues working with it until the next failure occurs or reverts to the primary Proxy refer to the Redundancy Mode parameter If none of the Proxy servers respond the gateway goes over the list again The gateway also provides real time switching hotswap mode between the primary and redundant proxies IsProxyHotSwap 1 If the first Proxy doesn t respond to INVITE message the same INVITE message is immediately sent
243. defined in the ini file or set from the Web browser 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 SIP User s Manual 22 Document LTRT 65408 SIP User s Manual 1 Overview The REGISTER message is sent to the Registrar s IP address if configured or to the Proxy s IP address The message is sent per gateway or per gateway endpoint according to the AuthenticationMode parameter Usually the FXS gateways are registered per gateway port while FXO gateways send a single registration message where Username is used instead of phone number in From To headers The registration request is resent according to the parameter RegistrartionTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the gateway resends its registration request after 3600 x 70 2520 sec The default value of RegistrartionTimeDivider is 50 Proxy and Registrar Authentication handling 401 and 407 responses using Basic or Digest methods Accepted challenges are kept for future requests to reduce the network traffic Single gateway Registration or multiple Registration of all gateway endpoints Supported methods INVITE CANCEL BYE ACK REGISTER OPTIONS INFO REFER UPDATE NOTIFY PRACK SUBSCRIBE and PUBLISH Modifying connection parameters for an already estab
244. demBypassCoderType The payload type used with these G 711 coders is a standard one 8 for G 711 A Law and 0 for G 711 u Law The parameters defining payload type for the old AudioCodes 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 0 There isn t a Cisco gateway at the remote side default 1 There is a Cisco gateway at the remote side When there is a Cisco gateway at the remote side the local gateway must set the value of the annexb parameter of the fmtp attribute in the SDP to no This logic should be used if EnableSilenceCompression 2 enable without adaptation In this case Silence Suppression should be used on the channel but not declared in the SDP Selects the Bellcore Caller ID sub standard 0 Between rings default 1 Not ring related 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 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 Selects the ET
245. dentity as an IP address and the update interval are configurable parameters that can be specified either in the ini file NTPServerlP NTPUpdatelnterval respectively or via an SNMP MIB object 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 should use is a configurable parameter that can be specified either in the ini file NTPServerUTCOffset or via an SNMP MIB object If required the clock update is performed by the client as the final step of the update process The update is done 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 in order 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 9 8 IP QoS via Differentiated Services DiffServ DiffServ is architecture providing different types or levels of service for IP traffic
246. der 105 NSE Bypass Version 5 0 361 December 2006 a WH wt AudioCodes MediaPack D 3 Default RTP RTCP T 38 Port Allocation The following table shows the default RTP RTCP T 38 port allocation Table D 3 Default RTP RTCP T 38 Port Allocation Channel Number RTP Port RTCP Port T 38 Port 1 6000 6001 6002 2 6010 6011 6012 3 6020 6021 6022 4 6030 6031 6032 5 6040 6041 6042 6 6050 6051 6052 7 6060 6061 6062 8 6070 6071 6072 9 6080 6081 6082 10 6090 6091 6092 11 6100 6101 6102 12 6110 6111 6112 13 6120 6121 6122 14 6130 6131 6132 15 6140 6141 6142 16 6150 6151 6152 17 6160 6161 6162 18 6170 6171 6172 19 6180 6181 6182 20 6190 6191 6192 21 6200 6201 6202 22 6210 6211 6212 23 6220 6221 6222 24 6230 6231 6232 Note To configure the gateway to use the same port for both RTP and T 38 packets set the parameter T38UseRTPPort to 1 SIP User s Manual 362 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools E Accessory Programs and Tools The accessory applications and tools shipped with the device provide you with friendly interfaces that enhance device usability and smooth your transition to the new VoIP infrastructure The following applications are available m TrunkPack Downloadable Conversion Utility refer to Section E 1 below E Call Progress Tones Wizard refer to Section E 1 3 on page 366 E 1 TrunkPack Downloadable Conversion Utility Use the TrunkPack Downloadable Conve
247. detailed information on the Ethernet interface configuration refer to Section 9 1 on page 247 gt To view the Ethernet Port Information parameters take this step E Open the Ethernet Port Information screen Advanced Configuration menu gt Network Settings gt Ethernet Port Information option the Ethernet Port Information screen is displayed Figure 5 60 Ethernet Port Information Screen Ethernet Port Information Port 1 Duplex Mode Port 1 Speed 100 mbps Table 5 61 Ethernet Port Information Parameters Parameter Description Port 1 Duplex Mode Shows the Duplex mode the Ethernet port is using Half Duplex or Full Duplex Port 1 Speed Shows the speed in Mbps that the Ethernet port is using 10 Mbps or 100 Mbps SIP User s Manual 194 Document LTRT 65408 SIP User s Manual 5 Web Management 5 8 Monitoring the MediaPack Channels Home Page The Channel Status screen provides real time monitoring on the current channels status In addition this screen allows you to assign a brief description or name to each port as well as releasing a channel N The Web interface provides the Home icon for quick and easy access to this screen 5 8 1 Viewing the Status of Channels gt To monitor the status of the MediaPack channels take this step E Open the Channel Status screen by clicking the Home icon the Channel Status screen is displayed different screen for FXS and FXO Fi
248. e 10 20 30 40 50 60 80 100 120 30 60 90 msec 10 20 30 40 50 60 80 100 120 10 20 30 40 50 60 msec Mute transfer in RTP payload or relay in compliance with RFC 2833 Dynamic range 0 to 25 dBm compliant with TIA 464B and Bellcore TR NWT 000506 32 tones single tone dual tones or AM tones programmable frequency amp amplitude 64 frequencies in the range 300 to 1980 Hz 1 to 4 cadences per tone up to 4 sets of ON OFF periods 32 dB to 31 dB in steps of 1 dB 32 dB to 31 dB in steps of 1 dB Group 3 fax relay up to 14 4 kbps with auto fallback T 38 compliant real time fax relay Tolerant network delay up to 9 seconds round trip Auto switch to PCM or ADPCM on V 34 or V 90 modem detection SIP RFC 3261 RTP RTCP packetization IP stack UDP TCP RTP Remote Software load TFTP HTTP and HTTPS Loop start Motorola PowerQUICC 870 SDRAM 32 MB AudioCodes AC482 VoIP DSP 2 4 or 8 Analog FXS phone or fax ports loop start RJ 11 4 or 8 Analog FXO PSTN PBX loop start ports MP 118 4 FXS 4 FXO ports MP 114 2 FXS 2 FXO ports 10 100 Base TX RS 232 Terminal Interface requires a DB 9 to PS 2 adaptor Channel status and activity LEDs The Lifeline provides a wired analog POTS phone connection to any PSTN or PBX FXS port when there is no power or the network fails Combined FXS FXO gateways provide a Lifeline connection available on all FXS ports Note The Lifeline splitte
249. e be 1 for the device to send device related initial startup information such as board type current IP address software version in the vendor specific information field in the BootP request This information can be viewed in the main screen of the BootP TFTP under column Client Info refer to Figure C 1 showing BootP TFTP main screen with the column Client Info on the extreme right For a full list of the vendor specific Information fields refer to Section 7 3 2 on page 214 Note This option is not available on DHCP servers Version 5 0 359 December 2006 7a Ta wt AudioCodes MediaPack C 12 Managing Client Templates Templates can be used to simplify configuration of clients when most of the parameters are the same v eh a eS OY ee y Figure C 5 Templates Screen Templates Template Name TemplateName e Pepp Subnet ss fas o fo Gatewy fo fs fo Po SewerlP fio fis fe fee BootFile faxxcmp INI File Jai x OK Apply To create a new template take these 4 steps Click the Add New Template button k l Fill in the default parameter values in the parameter fields Click Apply to save this new template Click OK when you are finished adding templates To edit an existing template take these 4 steps Select the template by clicking on its name from the list of templates in the window Make changes to the parameters as required Click Apply to save this new template Click OK
250. e encoded ini file option described in Section 6 1 on page 209 Note 3 After it is configured the value of the pre shared key cannot be obtained via Web ini file or SNMP refer to Section 12 1 3 3 on page 287 Determines the encryption type used in the main mode negotiation for up to four proposals X stands for the proposal number 0 to 3 The valid encryption values are Not Defined default DES CBC 1 Triple DES CBC 2 AES 3 Determines the authentication protocol used in the main mode negotiation for up to four proposals X stands for the proposal number 0 to 3 The valid authentication values are Not Defined default HMAC SHA1 96 2 HMAC MD5 96 4 Determines the length of the key created by the DH protocol for up to four proposals X stands for the proposal number 0 to 3 The valid DH Group values are Not Defined default DH 786 Bit 0 DH 1024 Bit 1 Version 5 0 281 December 2006 Ta fal AudioCodes MediaPack Table 12 1 IKE Table Configuration Parameters continues on pages 281 on 282 Parameter Name Authentication Method Description Determines the authentication method for IKE IkePolicyAuthenticationMeth The valid authentication method values include od IKE SA LifeTime sec IKEPolicyLifelnSec IKE SA LifeTime KB IKEPolicyLifelnKB 0 Pre shared Key default 1 RSA Signiture Note 1 For pre shared key based authentication peers partici
251. e 2 1 on page 26 The MP 124 front panel LEDs are described in Table 2 4 on page 27 the rear panel LEDs are described in Table 2 6 on page 28 E Self Testing on hardware initialization refer to Section 13 1 below E Error notification messages via the following interfaces e Syslog Log messages can be viewed using an external Syslog server refer to Section 13 2 on page 301 or on the Message Log screen in the Embedded Web Server refer to Section 5 7 2 on page 192 Note that the Message Log screen is not recommended for prolong debugging e RS 232 terminal For information on establishing a serial communications link with the MediaPack refer to Section 10 2 on page 262 Self Testing The MediaPack features two self testing modes used to identify faulty hardware components E Rapid The Rapid test is performed every time the MediaPack starts up It is executed each time the MediaPack completes its initialization process This is a short test phase in which the only error detected and reported is failure in initializing hardware components If an error is detected an error message is sent to the Syslog Detailed Used in addition to the Rapid and Enhanced test modes The test is performed on startup when initialization of the MediaPack is completed and if the parameter EnableDiagnostics is set to 1 or 2 In this mode the MediaPack tests its DSPs RAM and flash memory When EnableDiagnostics is set to 1 flash is tested
252. e 202 The prerecorded tones are played repeatedly This enables you to record only part of the tone and play it 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 Note The maximum size of a PRT file that can be loaded to the gateway is 100 KB SIP User s Manual 330 Document LTRT 65408 SIP User s Manual 15 Configuration Files 15 3 The Coefficient Configuration File The Coeff_FXS dat file is used to provide best termination and transmission quality adaptation for different line types for FXS gateways This adaptation is performed by modifying the telephony interface characteristics such as DC and AC impedance feeding current and ringing voltage The coeff dat configuration file is produced specifically for each market after comprehensive performance analysis and testing and can be modified on request The current file supports US line type of 600 ohm AC impedance and40 V RMS ringing voltage for REN 2 To load the coeff dat file to the MediaPack use the Embedded Web Server refer to Section 5 6 4 on page 166 or alternatively specify the FXS coeff dat file name in the gateway s ini file refer to Section 5 9 2 1 on page 203 The Coeff dat file consists of a set of parame
253. e enables the IP gateway 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 reception each coder receives its ptime value in the following precedence From mptime attribute From ptime attribute Default value Retransmission Parameters SIP T1 Retransmission Timer msec SipT1Rtx SIP T2 Retransmission Timer msec SipT2Rtx SIP Maximum Rtx SIPMaxRtx EnableReasonHeader Enable Reason Header 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 The maximum interval in msec between 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 Nu
254. e 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 gateway notices that the buffer is not filling up as much as expected At no time does the buffer decrease to less than the minimum size configured by the Minimum delay parameter Special Optimization Factor Value 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 so
255. e gateway and you don t want any traffic to interfere with the process gt To lock the gateway take these 4 steps 1 Open the Maintenance Actions screen Maintenance menu the Maintenance Actions screen is displayed Figure 5 71 Maintenance Actions Screen Reset Board Burn To FLASH Yes v No Graceful Option v LOCK UNLOCK Lock Timeout sec Current Admin State Save Configuration Save Configuration 2 Under the LOCK UNLOCK group from the Graceful Option drop down list select one of the following options e Yes The gateway 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 gateway is locked regardless of traffic Any existing traffic is terminated immediately 3 In the Lock Timeout field relevant only if the Graceful Option in the previous step is set to Yes enter the time in seconds after which the gateway locks Note that if no traffic exists and the time has not expired the gateway locks SIP User s Manual 204 Document LTRT 65408 SIP User s Manual 5 Web Management 4 Click the LOCK button 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 pro
256. e in a voice recording utility such as CoolEdit Locate the tone that the PBX played to indicate the disconnected call if such a tone exists Locate the attributes of the tone its frequency and interval on off time In the Call Progress Tones file add a new Reorder Tone with the attributes you found in the previous step Ensure that you update the numbers of the successive tones and the total number of tones in the beginning of the file Create a Call Progress Tones dat file using the DConvert Utility refer to Section E 1 on page 363 Load the new file to the gateway and then reset the gateway SIP User s Manual 374 Document LTRT 65408 SIP User s Manual F SNMP Traps F F 1 SNMP Traps This section provides information on proprietary SNMP traps currently supported by the gateway There is a separation between traps that are alarms and traps that are not logs Currently all have the same structure made up of the same 11 varbinds Variable Binding 1 3 6 1 4 1 5003 9 10 1 21 1 The source varbind is composed of a string that details the component from which the trap is being sent forwarded by the hierarchy in which it resides For example an alarm from an SS7 link has the following string in its source varbind acBoard 1 SS7 0 SS7Link 6 In this example the SS7 link number is specified as 6 and is part of the only SS7 module in the device that is placed in slot number 1 in a chassis and is the module
257. e number E666 k Cancel 2 Enter the IP address of the MediaPack FXO gateway you are using 3 Select the gateway s ports that are connected to your PBX and specify the phone number of each extension 4 In the Invalid phone number field enter a number that generates a fast busy tone when dialed Usually any incorrect phone number should cause a fast busy tone 5 Click Next The CPTWizard communicates with the FXO gateway via TPNCP TrunkPack Network Control Protocol If this protocol has been disabled in the gateway configuration the CPTWizard doesn t display the next screen and an error is reported SIP User s Manual 368 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools E 2 4 Recording Screen Automatic Mode After the connection to the MediaPack FXO gateway is established the recording screen is displayed Figure E 7 Recording Screen Automatic Mode AudioCodes Call Progress Tones Wizard 2 x Automatic Manual Automatic tone detection and analysis Start Automatic Configuration Status idle m Tones analyzed Tone Type LoFreq Hi Freg 1st On 1st Off 2nd On 2nd Off Detected c 3 Aud ioCodes i Cancel gt To start recording in automatic mode take these 3 steps 1 Click the Start Automatic Configuration button the wizard starts the following Call Progress Tones detection sequence the operation takes app
258. e 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 Continuous e g dial tone a steady non interrupted sound 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 m Cadence A repeating sequence of on and off sounds Up to four different sets of on off periods can be specified 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 Version 5 0 325 December 2006 a WH wt AudioCodes MediaPack Users 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 users 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 MediaPack 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
259. e the Embedded Web Server s Quick Setup screen shown in Figure 4 1 below Refer to Section 5 3 on page 51 for information on accessing the Quick Setup screen Figure 4 1 Quick Setup Screen IP Configuration IP Address 082580 NAT IP Address pooo Subnet Mask 255 255 0 0 0 Default Gateway IP Address hoso SIP Parameters Gateway Name j10 8 8 10 Working with Proxy Yes z Proxy IP Address 10 8 8 10 Enable Tables Coders Table gt Tel to IP Routing Table o gt Endpoint Phone Number Table _ gt gt To configure basic SIP parameters take these 9 steps Proxy Name Enable Registration 1 If the MediaPack is connected to a router with Network Address Translation NAT enabled perform the following procedure If it isn t leave the NAT IP Address field undefined e Determine the public IP address assigned to the router by using for instance router Web management If the public IP address is static enter this in the NAT IP Address field e Enable the DMZ Demilitarized Zone configuration on the router for the LAN port where the MediaPack gateway is connected This enables unknown packets to be routed to the DMZ port Under SIP Parameters enter the MediaPack domain name in the field Gateway Name If the field is not specified the MediaPack IP address is used instead default When working
260. e value of a named object Get Next A request that returns the next name and value of the next object supported by a network device given a valid SNMP name E Set A request that sets a named object to a specific value E Trap A message generated asynchronously by network devices It is an unsolicited message from an agent to the manager Each of these message types fulfills a particular requirement of Network Managers E Get Request Specific values can be fetched via the get request to determine the performance and state of the device Typically many different values and parameters can be determined via SNMP without the overhead associated with logging into the device or establishing a TCP connection with the device Get Next Request Enables the SNMP standard network managers to walk through all SNMP values of a device via the get next request to determine all names and values that an operant device supports This is accomplished by beginning with the first SNMP object to be fetched fetching the next name with a get next and repeating this operation Version 5 0 305 December 2006 Ta WH wt AudioCodes MediaPack m Set Request The SNMP standard provides a method of effecting an action associated with a device via the set request to accomplish activities such as disabling interfaces disconnecting users clearing registers etc This provides a way of configuring and control
261. eb Management Table 5 44 Media Settings RTP RTCP Parameters continues on pages 158 to 159 Parameter RTP Base UDP Port BaseUDPPort Remote RTP Base UDP Port RemoteBaseUDPPort RTP Multiplexing Local UDP Port L1L1ComplexTxUDPPort RTP Multiplexing Remote UDP Port L1L1ComplexRxUDPPort Comfort Noise Generation Negotiation ComfortNoiseNegotiation Description Lower boundary of UDP port used for RTP RTCP Real Time Control Protocol RTP port 1 and T 38 RTP port 2 The upper boundary is the Base UDP Port 10 number of gateway 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 the default then The first channel uses the following ports RTP 6000 RTCP 6001 and T 38 6002 the second channel uses RTP 6010 RTCP 6011 and T 38 6012 etc Note 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 Section D 3 on page 362 Determines the lower boundary of UDP ports used for RTP RTCP and T 38 by a remote gateway If this parameter is set to a non zero value ThroughPacket is enabled Note that the value of RemoteBaseUDPPort on the local gateway must equal the value of BaseUDPPort of the remote gateway The gateway uses these parameters to identify a
262. econd 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 328 Document LTRT 65408 SIP User s Manual 15 Configuration Files 15 1 2 1 Examples of Various Ringing Signals Figure 15 4 Examples of Various Ringing Signals 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 5 0 329 December 2006 Ta W
263. ecovered using the Web browser The BootP is normally used to configure the device s initial parameters Once this information has been provided the BootP is no longer needed All parameters are stored in non volatile memory and used when the BootP is not accessible C 2 An Overview of BootP BootP is a protocol defined in RFC 951 and RFC 1542 that enables an internet device to discover its own IP address and the IP address of a BootP on the network and to obtain the files from that utility that need to be loaded into the device to function A device that uses BootP when it powers up broadcasts a BootRequest message on the network A BootP on the network receives this message and generates a BootReply The BootReply indicates the IP address that should be used by the device and specifies an IP address from which the unit may load configuration files using Trivial File Transfer Protocol TFTP described in RFC 906 and RFC 1350 C 3 Key Features Internal BootP supporting hundreds of entities Internal TFTP Contains all required data for our products in predefined format Provides a TFTP address enabling network separation of TFTP and BootP utilities Tools to backup and restore the local database Templates User defined names for each entity Option for changing MAC address Protection against entering faulty information Remote reset Version 5 0 349 December 2006 a WH wt AudioCodes MediaPack Unicast BootP re
264. ect to this Access Concentrator If this initial connection succeeds then the PPP LCP phase starts each side of the PPPoE connection sends LCP configuration requests to configure the PPP link The gateway PPPoE client supports both PAP and CHAP authentications The type of authentication protocol used is according to the request from the authentication server In the LCP configuration phase the server requires a specific authentication none PAP or CHAP are supported The ini file parameters PPPoEUserName PPPoEPassword and PPPoEServerName are used to configure the authentication parameters If the Access Concentrator is configured to operate in PAP the PPPoEUserName and PPPoEPassword are used as Username and Password in this case the PPPoEServerName parameter is not used If the Access Concentrator is configured to operate in CHAP the PPPoEUserName parameter functions as Client Name sent in the CHAP response packet while the PPPoEPassword functions as the shared secret calculated along with the challenge to produce the response In this case the PPPoEServerName is the name of the server Some hosts can be configured to authenticate to multiple servers In such hosts the server name is used to identify the secret that should be used Note The AudioCodes gateway being a PPPoE client requests no authentication After the gateway has been authenticated it needs to configure a network layer protocol The gateway uses the IP protocol
265. ed Enter coders in the following format CoderName lt Coder Name gt lt Ptime gt lt Rate gt lt Payload Type gt lt Silence Suppression Mode gt Note 1 The coder name is case sensitive Note 2 If silence suppression is not defined for a specific coder the value defined by the parameter EnableSilenceCompression is used Note 3 The value of several fields is hard coded according to well known standards e g the 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 For example CoderName g711Alaw64k 20 0 CoderName g 11Ulaw64k 40 CoderName g7231 90 1 1 CoderName g726 0 Notes For an explanation on V 152 support and implementation of T 38 and VBD coders refer to Section 8 3 5 on page 222 Version 5 0 T3 December 2006 Ta Ao wl AudioCodes MediaPack 5 5 1 4 DTMF amp Dialing Parameters Use this screen to configure parameters that are associated with DTMF and dialing gt To configure the dialing parameters take these 4 steps 1 Open the DTMF amp Dialing screen Protocol Management menu gt Protocol Definition submenu gt DTMF amp Dialing option the DTMF amp Dialing screen is displayed Figure 5 8 DTMF amp Dialing Screen DTMF amp Dialing Max Digits In Phone Num Int
266. ed in the Access Level drop down list select the new access level and click the button Change Access Level the new access level is applied immediately 3 To change the username of an account enter the new username in the field User Name and click the button Change User Name the new username is applied immediately and the Enter Network Password screen appears shown in Figure 5 1 on page 51 Enter the updated username in the Enter Network Password screen Note that the username can be a maximum of 19 case sensitive characters 4 To change the password of an account enter the current password in the field Current Password the new password in the fields New Password and Confirm New Password and click the button Change Password the new password is applied immediately and the Enter Network Password screen appears shown in Figure 5 1 on page 51 Enter the updated password in the Enter Network Password screen Note that the password can be a maximum of 19 case sensitive characters A user with a Security Administrator access level can change all attributes for all accounts Users with an access level other than Security Administrator can only change their own password and username Version 5 0 169 December 2006 7a Ta wt AudioCodes MediaPack 5 6 5 2 Configuring the Web and Telnet Access List Use this screen to define up to ten IP addresses that are permitted t
267. ed Note that if DisconnectOnBusyTone 0 the FXO gateway ignores the detection of Busy Reorder tones and doesn t release the call For all other MediaPack 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 exist or if the phone number isn t found the MediaPack sends 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 SIP 404 response By changing its value to 34 SIP 503 response is sent Other causes can be used as well Call Detail Record The Call Detail Record CDR contains vital statistic information on calls made by the gateway CDRs are generated at the end and optionally at the beginning of each call determined by the parameter CDRReportLevel and sent to a Syslog server The destination IP address for CDR logs is determined by the parameter CDRSyslogServerIP The table below lists the CDR fields that are supported Table 8 1 Supported CDR Fields continues on pages 232 to 233 Field Name Description Cid Port Number Callld H 323 SIP Call Identifier Trunk N A BChan N A Conld H 323 SIP Conference ID TG Trunk Group Number EPTyp Endpoint Type Orig Call Originator IP Tel Sourcelp Source IP Address Destlp Destination IP Address TON Source Phone Number Type NPI Source Phone
268. ed default value to it 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 in exactly the same order A line in a table is identified by its table name and Index fields Each such line may appear only once in the ini file 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 SIP User s Manual 270 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 6 Customizing the MediaPack Web Interface Customers incorporating the MediaPack into their portfolios can customize the Web Interface to suit their specific corporate logo and product naming conventions Customers can customize the Web Interface s title bar AudioCodes title bar is shown in Figure 10 5 a customized title bar is shown in Figure 10 7 Figure 10 5 User Customizable Web Interface Title Bar ta AudioCodes Product Name Figure 10 6 Customized Web Interface Title Bar idgets Inc tel la To customize the title bar via the Web Interface take th
269. ed Web Server refer to Section 5 6 3 on page 165 Version 5 0 261 December 2006 Ta WH wt AudioCodes MediaPack 10 2 Establishing a Serial Communications Link with the MediaPack Use serial communication software e g HyperTerminal to establish a serial communications link with the MediaPack via the RS 232 connection You can use this link to change the networking parameters Section 4 2 4 on page 44 and to receive error notification messages gt To establish a serial communications link with the MediaPack via the RS 232 port take these 2 steps 1 Connect the RS 232 port to your PC For the MP 124 refer to Section 3 2 4 1 on page 40 For the MP 11x refer to Section 3 1 5 1 on page 34 2 Use a serial communication software e g HyperTerminal with the following communications port settings e Baud Rate 115 200 bps MP 124 9 600 bps MP 11x e Databits 8 e Parity None e Stopbits 1 e Flow control None Note that after resetting the gateway the information shown in Figure 11 1 below appears on the terminal screen This information can be used to determine possible MediaPack initialization problems such as incorrectly defined or undefined local IP address subnet mask etc Figure 10 1 RS 232 Status and Error Messages MAC address 00 90 8F 01 00 9E Local IP address 10 1 37 6 Subnet mask 255 255 0 0 Default gateway IP address 10 1 1 5 TFTP server IP address 10 1
270. ediaPack Table 5 12 Keypad Features Parameters Parameter Description Forward Note that the forward type and number can be viewed in the Call Forward Table refer to Section 5 5 9 5 on page 124 Enconaitional Keypad sequence that activates the immediate forward option KeyCFUnCond i Caen nswer Keypad sequence that activates the forward on no answer option eve ied Keypad sequence that activates the forward on busy option On Busy or No Answer KeyCFBusyOrNoAnswer Do Not Disturb Keypad sequence that activates the Do Not Disturb option immediately reject KeyCFDoNotDisturb incoming calls To activate the required forward method from the telephone Dial the preconfigured sequence number on the keypad a dial tone is heard Dial the telephone number to which the call is forwarded terminate the number with a confirmation tone Keypad sequence that activates the forward on busy or no answer option is heard Deactivate Keypad sequence that deactivates any of the forward options KeyCFDeact After the sequence is pressed a confirmation tone is heard Caller ID Restriction Note that the caller ID presentation can be viewed in the Caller Display Information table refer to Section 5 5 9 3 on page 121 Activate Keypad sequence that activates the restricted Caller ID option KeyCLIR After the sequence is pressed a confirmation tone is heard Deactivate Keypad sequence that deactivates the restricted Caller ID
271. efault 1 1 Flow debugging is enabled 2 2 Flow and device interface debugging are enabled 3 3 Flow device interface and stack interface debugging are enabled 4 4 Flow device interface stack interface and session manager debugging are enabled 5 5 Flow device interface stack interface session manager and device interface expanded debugging are enabled Note Usually set to 5 if debug traces are needed No PI 0 For Tel gt IP calls the gateway sends 180 Ringing SIP response to IP after placing a call to phone FXS or to PBX FXO PI 1 PI 8 1 8 For TelSIP calls if EnableEarlyMedia 1 the gateway sends 183 session in progress message SDP immediately after a call is placed to Phone PBX This is used to cut through the voice path before remote party answers the call enabling the originating party to listen to network Call Progress Tones such as Ringback tone or other network announcements Not Configured 1 Default values are used The default for FXO gateways is 1 The default for FXS gateways is 0 SIP User s Manual 80 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 8 Advanced Parameters General Parameters continues on pages 78 to 82 Parameter Enable Busy Out EnableBusyOut Default Release Cause DefaultReleaseCause Delay After Reset sec GWAppDelayTime Max Number of Active Calls MaxActiveCalls Max Call Duratio
272. el The figure below illustrates the rear panel of the MP 124 For descriptions of the rear panel components refer to Table 2 5 For the functionality of rear panel Ethernet LEDs refer to Table 2 6 Figure 2 4 MP 124 FXS Rear Panel Connectors Table 2 5 MP 124 Rear Panel Component Descriptions Item Label Component Description 1 l Protective earthing screw mandatory for all installations Accepts a 6 32 UNC screw 2 100 250 V AC power supply socket 50 60 Hz 2A 3 ANALOG FXS LINES 50 pin Telco for 1 to 24 analog lines 1 24 4 RS 232 9 pin RS 232 status port 5 ETHERNET 10 100 Base TX Ethernet connection The Ethernet LEDs are located within the RJ 45 socket The table below describes these LEDs Table 2 6 Ethernet LEDs on the MP 124 Rear Panel Label Type Color State Function ETHERNET Ethernet Status Green On Valid 10 100 Base TX Ethernet connection Red On Malfunction SIP User s Manual 28 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack 3 Installing the MediaPack This section provides information on the installation procedure for the MP 11x refer to Section 3 1 below and the MP 124 refer to Section 3 2 on page 36 For information on how to start using the gateway refer to Chapter 4 on page 41 Caution Electrical Shock The equipment must only be installed or serviced by qualified service personnel 3 1 Installing the MP 11x To install the MP 11x take these 4 steps
273. elected port in the format lt IP Address gt lt port gt Note 1 This parameter is available only if you select Use Proxy in the Enable Proxy field Note 2 When port number is specified DNS NAPTR SRV queries aren t performed even if ProxyDNSQueryType is set to 1 ini file note The IP addresses of the third redundant Proxy is defined by the fourth repetition of the ini file parameter ProxylP Version 5 0 67 December 2006 Ta WH wt AudioCodes MediaPack Table 5 4 Proxy amp Registration Parameters continues on pages 66 to 71 Parameter Proxy Load Balancing Method ProxyLoadBalancingMe thod Proxy IP List Refresh Time ProxylPListRefreshTim e DNS Query Type DNSQueryType Enable SRV Queries EnableSRVQuery Description Enables the usage of the Proxy Load Balancing mechanism Valid options include Disable 0 Load Balancing is disabled default Round Robin 1 Round Robin algorithm Random Weights 2 Random Weights When Round Robin 1 algorithm is used a list of all possible Proxy IP addresses is compiled This list includes all entries in the ProxylP table after necessary DNS resolutions including NAPTR and SRV if configured This list can handle up to 15 entries After this list is compiled the Proxy Keep Alive mechanism according to EnableProxyKeepAlive and ProxyKeepAliveTime is used to mark each entry as Offline or Online The balancing is only
274. ember 2006 Ta r AudioCodes MediaPack Figure 5 66 Load an ini File Screen CMP file Load an ini file from your computer to the device Browse Use existing configuration Device will revert to default configuration if no configuration is chosen 7 In the Load an ini File screen you can now choose to either e Click Browse and navigate to the ini file the check box Use existing configuration by default checked becomes unchecked Click Send File the ini file is loaded to the MediaPack and you re notified as to a successful loading e Ignore the Browse button its field remains undefined and the check box Use existing configuration remains checked by default e Ignore the Browse button and uncheck the Use existing configuration check box no ini file is loaded the MediaPack uses its factory preconfigured values 8 Youcan now choose to either e Click Cancel the MediaPack resets utilizing the cmp ini and all other configuration files that were previously stored in flash memory Note that these are NOT the files you loaded in the previous Wizard steps e Click Reset the MediaPack resets utilizing the new cmp and ini file you loaded up to now as well as utilizing the other configuration files e Click Back the Load a cmp file screen is reverted to refer to Figure 5 64 e Click Next the Load a CPT File screen opens refer to Figure 5 67 Loading a new CPT file
275. en the Application Settings screen Advanced Configuration menu gt Network Settings gt Application Settings option the Application Settings screen is displayed Figure 5 36 Application Settings Screen Application Settings NTP Settings NTP Server IP Address 0 0 0 0 NTP UTC Offset Hours fo Minutes fo NTP Update Interval Hours 24 Minutes fo Telnet Settings EmbBdded Telnet Server Disable Telnet Server TCP Port Telnet Server Idle Timeout STUN Settings Enable STUN Disable STUN Server Primary IP 0 0 0 0 STUN Server Secondary IP 0 0 0 0 NFS Settings 2 Configure the Application Settings according to Table 5 37 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Version 5 0 141 December 2006 Ta fal AudioCodes MediaPack Table 5 37 Network Settings Application Settings Parameters Parameter NTP Settings Description For detailed information on NTP refer to Section 9 7 on page 253 NTP Server IP Address NTPServerIP NTP UTC Offset NTPServerUTCOffset NTP Update Interval NTPUpdatelnterval Telnet Settings Embedded Telnet Server TelnetServerEnable Telnet Server TCP Port TelnetServerPort Telnet Server Idle Timeout TelnetServerldleDisconnect STUN Settings Enable STUN EnableSTUN STUN Server Primary IP STUNServerPrimaryIP
276. enabled SIP INVITE message is first sent to the primary Proxy server If there is no response from the primary Proxy server for Number of RTX before Hot Swap retransmissions the INVITE message is resent to the redundant Proxy server Number of retransmitted INVITE messages before call is routed hot swapped to another Proxy 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 table If a domain name in the routing table is resolved into 2 IP addresses and if there is no response for ProxyHotSwapRtx retransmissions to the INVITE message that is sent to the first IP address the gateway immediately initiates a call to the second IP address Username used for Registration and for Basic Digest authentication process with Proxy Registrar Parameter doesn t have a default value empty string Note Applicable only if single gateway registration is used Authentication Mode Authentication Per gateway Password used for Basic Digest authentication process with Proxy Registrar Single password is used for all gateway ports The default is Default_Passwd Note The Authentication table can be used instead String used by the server and client to provide mutual authentication Free format i e Cnonce 0a4f113b The default is Default_Cnonce Per Endpoint 0 Registration amp Authentication separately for e
277. ent LTRT 65408 SIP User s Manual 5 Web Management 5 5 2 4 Keypad Features The Keypad Features screen applicable only to FXS gateways enables you to activate deactivate the following features directly from the connected telephone s keypad ae Hotline refer to Section 5 5 9 2 on page 120 Caller ID Restriction refer to Section 5 5 9 3 on page 121 Call Forward refer to Section 5 5 9 4 on page 122 To configure the keypad features take these 4 steps Open the Keypad Features screen Protocol Management menu gt Advanced Parameters submenu gt Keypad Features option the Keypad Features screen is displayed Figure 5 13 Keypad Features Screen Keypad Features Unconditional No Answer On Busy On Busy or No Answer Do Not Disturb Deactivate Caller ID Restriction Activate Deactivate Activate Deactivate Configure the Keypad Features according to Table 5 12 Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 The method used by the gateway 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 it is not per call Version 5 0 89 December 2006 Ta WH wt AudioCodes M
278. ent unauthorized access to the Embedded Web Server two user accounts are available a primary and secondary Each account is composed of three attributes username password and access level For detailed information on the user account mechanism refer to Section 5 2 1 on page 49 It is recommended that you change the default username and password of the account you use to access the Embedded Web Server SIP User s Manual 168 Document LTRT 65408 SIP User s Manual 5 Web Management gt To change the Web User Accounts attributes take these 4 steps 1 Open the Web User Accounts screen Advanced Configuration menu gt Security Settings gt Web User Accounts option the Web User Accounts screen is displayed Figure 5 47 Web User Accounts Screen for Users with Security Administrator Privileges Current Logged User Admin Account Data for User Admin User Name Admin Access Level Security Administrator Fill in the following 3 fields to change the password Current Password New Password Confirm New Password Account Data for User 8 User Name 8 Access Level User Monitor v Fill in the following 3 fields to change the password Current Password New Password Confirm New Password 2 To change the access level of the secondary account the access level of the primary account cannot be chang
279. enter the IP address and optionally port number Profile ID Status that is assigned to these prefixes Domain names such as domain com can be used instead of IP addresses For example lt IP Address gt lt Port gt To discard outgoing IP calls enter 0 0 0 0 in this field Note When using domain names you must enter a DNS server IP address or alternatively define these names in the Internal DNS Table Enter the number of the IP profile that is assigned to the destination IP address defined in the Destination IP Address field A read only field representing the quality of service of the destination IP address N A Alternative Routing feature is disabled OK IP route is available Ping Error No ping to IP destination route is not available QoS Low Bad QoS of IP destination route is not available DNS Error No DNS resolution only when domain name is used instead of an IP address Charge Code An optional Charge Code 1 to 25 can be applied to each routing rule to associate it with an entry in the Charge Code table refer to Section 5 5 2 3 1 on page 88 Parameter Name in ini File Parameter Format Prefix 5 5 5 3 Prefix lt Destination Phone Prefix gt lt IP Address gt lt Src Phone Prefix gt lt IP Profile ID gt lt Charge Code gt For example Prefix 20 10 2 10 2 202 1 15 Prefix 10 340 451 xxx 10 2 10 6 1 1 Prefix gateway domain com 20 Note 1 lt destination so
280. eps 1 Select the radio button of the entry you want to activate 2 Click the DeActivate Rule button the rule is de activated Version 5 0 171 December 2006 F MediaPack a C wt AudioCodes gt 1 2 3 page 205 Parameter Is Rule Active Source IP AccessList_Source_IP Mask AccessList_Net_Mask 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_Burst Action Upon Match AccessList_Allow_Type Match Count AccessList_MatchCount To delete a rule take these 3 steps Select the radio button of the entry you want to activate Click the Delete Rule button the rule is deleted To save the changes so they are available after a power fail refer to Section 5 10 2 on Table 5 49 Internal Firewall Fields Description A read only field that indicates whether the rule is active or not Note After reset all rules are active IP address or DNS name of source network or a specific host IP network mask 255 255 255 255 for a single host or the appropriate value for the source IP addresses The IP address of the sender of the incoming packet is bitwise ANDed with this mask and then compared to the field 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 protoc
281. er Refer Replaces refer to Section 8 1 3 on page 216 Call Forward 3xx Redirect Responses refer to Section 8 1 4 on page 217 Call Waiting 182 Queued Response refer to Section 8 1 5 on page 217 Message Waiting Indication MWI refer to Section 8 1 6 on page 218 To activate these supplementary services Hold Transfer Forward Waiting and MWI on the MediaPack gateway enable each service s corresponding parameter either from the Web Interface or via the ini file Note that all call participants must support the specific used method When working with application servers such as BroadSoft s BroadWorks in client server mode the application server controls all supplementary services and keypad features by itself the gateway s supplementary services must be disabled Call Hold and Retrieve Initiating Hold Retrieve 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 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 hook flash After a successful retrieve voice is connected again Hold is performed by sending a REINVITE with the IP address 0 0 0 0 or a sendonly in the SDP according to the parameter HoldFormat
282. er Digit Timeout for Overlap Dialing sec Declare RFC 2853 in SDP 1st Tx DTMF Option RFC 2833 2nd Tx DTMF Option Not Supported 3rd Tx DTMF Option Not Supported 4th Tx DTMF Option g Not Supported 5th Tx DTMF Option Not Supported RFC 2833 Payload Type 96 Hook Flash Option a Not Supported Digit Mapping Rules Dial Tone Duration sec 16 Hotline Dial Tone Duration sec 16 Enable Special Digits Disable Default Destination Number 1000 2 Configure the DTMF amp Dialing parameters according to Table 5 7 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 7 DTMF amp Dialing Parameters continues on pages 74 to 76 Parameter Description Max Digits in Phone Num Maximum number of digits that can be dialed MaxDigits The valid range is 1 to 49 The default value is 5 Note Digit Mapping Rules Note Dialing ends when the maximum number of digits is dialed the Interdigit can be used instead Timeout expires the key is dialed or a digit map pattern is matched Inter Digits Timeout for Time in seconds that the gateway waits between digits dialed by the user When Overlap Dialing sec the Interdigit Timeout expires the gateway attempts to dial the digits already TimeBetweenDigits received The valid ran
283. er 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 gateway uses a jitter buffer The jitter buffer collects voice packets stores them and sends them to the voice processor in evenly spaced intervals The MediaPack uses a dynamic jitter buffer that can be configured using two parameters 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 gateway always buffers incoming packets by at least 10 msec worth of voice frames E 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 decays back down 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 th
284. er in the trap s OID For example the name of acBoardEthernetLinkAlarm is 9 The OID for acBoardEthernetLinkAlarm is 1 3 6 1 4 1 5003 9 10 1 21 2 0 10 Version 5 0 381 December 2006 7 T E MediaPack Reader s Notes SIP User s Manual 382 Document LT RT 65408 SIP User s Manual G Installation and Configuration of Apache HTTP Server G Installation and Configuration of Apache HTTP Server This appendix describes the installation and configuration of Apache s HTTP server with Perl script environment required for recording G 1 Windows 2000 XP Operation Systems Note For detailed installation information refer to http perl apache org docs 2 0 os win32 config html Additional required software an uploading script put cgi supplied with the software package gt To configure the Apache HTTP server and mod_perl version 2 0 software take these 9 steps 1 Download the third party Perl 5 8 win32 bin exe installation file from the following link www apache orq dist perl win32 bin Perl 5 8 win32 bin exe The installation file includes Apache 2 0 46 Perl 5 8 0 and mod_perl 1 99 the content of the file and the software version are subject to modification and changes in the future For full installation instructions refer to www apache org dist perl win32 bin Perl 5 8 win32 bin readme 2 To start the installation wizard run the Perl 5 8 win32 bin exe f
285. er s Manual 364 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools E 1 2 Encoding Decoding an ini File For detailed information on secured ini file refer to Section 6 1 on page 209 gt 1 To encode an ini file take these 6 steps Execute the TrunkPack Downloadable Conversion Utility DConvert exe supplied with the software package the utility s main screen opens shown in Figure E 1 Click the Process Encoded Decoded ini file s button the Encode Decode ini File s screen shown in Figure E 3 opens Figure E 3 Encode Decode ini File s Screen Encode Decode ini File s m Encode ini File s Input File s C Media Gateway gu_fxo ini Output le media gateway gqu_fxo aen F Use password Password Select File Encode File s m Decode ini File s Input File s CMedia Gateway gu_fxo aen Output Je media gatewaygw_fxo_dl ini Decode File s nmm y 4 Click the Select File button under the Encode ini File s section Navigate to the folder that contains the ini file you want to encode Click the ini file and click the Open button the name and path of both the ini file and the output encoded file appear in the fields under the Select File button Note that the name and extension of the output file can be modified Click the Encode File s button an encoded ini file with the name and extension you specified is created
286. erface disable the Web server DisableWebTask If you use Telnet do not use the default port 23 Use SSL mode to protect Telnet traffic from network sniffing If you use SNMP do not leave the community strings at their default values as they can be easily guessed by hackers refer to Section 14 8 1 on page 313 Use a firewall to protect your VoIP network from external attacks Network robustness may be compromised if the network is exposed to Denial of Service DoS attacks DoS attacks are mitigated by Stateful firewalls Do not allow unauthorized traffic to reach the MediaPack Legal Notice By default the MediaPack supports export grade 40 bit and 56 bit encryption due to US government restrictions on the export of security technologies To enable 128 bit and 256 bit encryption on your device contact your AudioCodes representative This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit www openssl org This product includes cryptographic software written by Eric Young eay cryptsoft com SIP User s Manual 300 Document LTRT 65408 SIP User s Manual 13 Diagnostics 13 13 1 13 2 Diagnostics Several diagnostic tools are provided enabling you to identify correct functioning of the MediaPack or an error condition with a probable cause and a solution or workaround m Front and rear panel LEDs on the MediaPack The MP 11x front panel LEDs are described in Tabl
287. ers The registration request is resent according to the parameter RegistrartionTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the gateway resends its registration request after 3600 x 70 2520 sec The default value of RegistrartionTimeDivider is 50 REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z9hG4bRaC7AU234 From lt sip 101 sipgatewayname gt tag 1c29347 To lt sip 101 sipgatewayname gt Calil wDpg IOASSEO2U2 179 22 229 Seq 1 REGISTER Expires 3600 Coniealciteaswesiso KONCZA eRe 22 Content Length 0 The servername string is defined according 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 sipgatewayname parameter defined in the ini file or set from the Web browser 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 SIP User s Manual 236 Document
288. ers requiring large deployments multiple media gateways in globally distributed enterprise offices for example that need to be managed by central personnel The EMS is not included in the device s supplied package Contact AudioCodes for detailed information on AudioCodes EMS and on AudioCodes EVN Enterprise VoIP Network solution for large VoIP deployments SIP User s Manual 324 Document LTRT 65408 SIP User s Manual 15 Configuration Files 15 Configuration Files This section describes the configuration dat files that are loaded in addition to the ini file to the gateway The configuration files are Call Progress Tones file refer to Section 15 1 on page 325 m Prerecorded Tones file refer to Section 15 2 on page 330 E FXS Coefficient file refer to Section 15 3 on page 331 m User Information file refer to Section 15 4 on page 332 To load either of the configuration files to the MediaPack use the Embedded Web Server refer to Section 5 9 2 on page 202 or alternatively specify the name of the relevant configuration file in the gateway s ini file and load it the ini file to the gateway refer to Section 5 9 2 1 on page 203 15 1 Configuring the Call Progress Tones and Distinctive Ringing File The Call Progress Tones and Distinctive Ringing configuration file used by the MediaPack is a binary file with the extension dat that is comprised of two sections The first section contains the def
289. es describing NFS failures or success FTP failure or success SIP User s Manual 380 Document LTRT 65408 SIP User s Manual F SNMP Traps F 1 5 Other Traps The following are provided as SNMP traps and are not alarms Trap Name OID MIB Note Trap Name OID MIB Trap Name OID MIB Severity Event Type Probable Cause Table F 14 coldStart Trap coldStart 1 3 6 1 6 3 1 1 5 1 SNMPv2 MIB This is a trap from the standard SNMP MIB Table F 15 authenticationFailure Trap authenticationFailure 1 3 6 1 6 3 1 1 5 5 SNMPv2 MIB Table F 16 acBoardEvBoardStarted Trap acBoardEvBoardStarted 1 3 6 1 4 1 5003 9 10 1 21 2 0 4 AcBoard cleared equipmentAlarm Other 0 Alarm Text Initialization Ended Note This is the AudioCodes Enterprise application cold start trap F 1 6 Trap Varbinds Each trap described above provides the following fields known as varbinds Refer to the AcBoard MIB for additional details on these varbinds acBoardTrapGlobalsName acBoardTrapGlobalsT extualDescription acBoardTrapGlobalsSource acBoardTrapGlobalsSeverity acBoardTrapGlobalsUniqID acBoardTrapGlobalsType acBoardTrapGlobalsDateAndTime acBoardTrapGlobalsProbableCause acBoardTrapGlobalsAdditionallnfo1 acBoardTrapGlobalsAdditionallnfo2 acBoardTrapGlobalsAdditionallnfo3 Note that acBoardTrapGlobalsName is actually a number The value of this varbind is X minus 1 where X is the last numb
290. escription CmpFileURL Specifies the name of the cmp file and the location of the server IP address or FQDN from which the gateway 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 Note 1 When this parameter is set in the ini file the gateway always loads the cmp file after it is reset Note 2 The cmp file is validated before it is burned to flash The checksum of the cmp file is also compared to the previously burnt checksum to avoid unnecessary resets Note 3 The maximum length of the URL address is 99 characters SIP User s Manual 184 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 56 Automatic Updates Parameters continues on pages 184 to 185 ini File Parameter Name IniFileURL PrtFileURL CptFileURL FXSCoeffFileURL UserInfoFileURL AutoUpdateCmpFile AutoUpdateFrequency AutoUpdatePredefinedTime ResetNow Description Specifies the name of the ini file and the location of the server IP address or FQDN from which the gateway loads the ini file 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 lt IP address gt lt file name gt Note 1 When using HTTP or HTTPS the date and time of the ini file are validated Only more recently dated
291. ese 3 steps Replace the main corporate logo refer to Section 10 6 1 below Replace the title bar s background image file refer to Section 10 6 2 on page 274 Customize the product s name refer to Section 10 6 3 on page 275 ON gt V 10 6 1 Replacing the Main Corporate Logo The main corporate logo can be replaced either with a different logo image file refer to Section 10 6 1 1 below or with a text string refer to Section 10 6 1 2 on page 273 Note that when the main corporation logo is replaced AudioCodes logo on the left bar refer to Figure 5 2 and in the Software Upgrade Wizard Section 5 9 1 on page 197 disappear Also note that the browser s title bar is automatically updated with the string assigned to the WebLogoText parameter when AudioCodes default logo is not used Version 5 0 271 December 2006 Ta WH wi AudioCodes MediaPack 10 6 1 1 Replacing the Main Corporate Logo with an Image File Use a gif jpg or jpeg file for the logo image It is important that the image file has a fixed height of 59 pixels the width can be configured up to a maximum of 339 pixels The size of the image files logo and background is limited each to 64 kbytes gt To replace the default logo with your own corporate image via the Web Interface take these 7 steps Access the MediaPack Embedded Web Server refer to Section 5 3 on page 51 2 Inthe URL field append the suffix AdminPage n
292. esets reverts to the initial value configured by RadiusLocalCacheTimeout Defines the time in seconds the locally stored username and password verified by the RADIUS server are valid When this time expires the username and password becomes invalid and a must re verified with the RADIUS server The valid range is 1 to OxFFFFFF 1 Never expires 0 Each request requires RADIUS authentication The default value is 300 5 minutes Defines the vendor ID the gateway 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 in the Vendor Specific Attributes VSA section of the received RADIUS packet The valid range is 0 to 255 The default value is 35 N A Enables or disables the Secure Real Time Transport Protocol SRTP Disable TGCP 0 SRTP is disabled default Enable SRTP 1 SRTP is enabled Note Use of 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 Determines the gateway s mode of operation when SRTP is used EnableMediaSecurity 1 Prefer 0 The gateway 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 Must 1 The gateway initiates encrypted calls If
293. ession is activated Sent packets are a duplication of the previously sent frame including duplication of the sequence number IP Multicasting The gateway supports IP Multicasting level 1 according to RFC 2236 i e IGMP version 2 for RTP channels The gateway is capable of transmitting and receiving Multicast packets Version 5 0 249 December 2006 a WH wt AudioCodes MediaPack 9 4 9 4 1 Point to Point Protocol over Ethernet PPPoE PPPoE is a method of sending the Point to Point Protocol over Ethernet network Point to Point Protocol PPP Overview Point to Point Protocol PPP provides a method of transmitting data over serial point to point links The protocol defines establishing configuring and testing the data link connection and the network protocol The PPP standard describes a state machine used to establish a valid connection between two hosts over a serial connection There are three major stages described helping to establish a network layer such as an IP connection over the point to point link LCP Link Configuration Protocol Authentication and NCP Network Control Protocol Once the network protocol is configured the two hosts can communicate sending network layer protocol such as IP over the PPP connection a small PPP header is added at the beginning of each packet At the initial phase the hosts use LCP link configuration protocol to negotiate for link characteristic and parameter
294. estination IP Address field in the Tel to IP Routing table for a match with the outgoing call Only if a match is not found a Proxy is used Possible uses for Tel to IP Routing can be as follows Can fallback to internal routing table if there is no communication with the Proxy servers Call Restriction when Proxy isn t used reject all outgoing Tel gt IP calls that are associated with the destination IP address 0 0 0 0 E P Security When the IP Security feature is enabled SecureCallFromlP 1 the VoIP gateway accepts only those IP gt Tel calls with a source IP address identical to one of the IP addresses entered in the Tel to IP Routing Table E Filter Calls to IP When a Proxy is used the gateway checks the Tel gt IP routing table before a telephone number is routed to the Proxy If the number is not allowed number isn t listed or a Call Restriction routing rule was applied the call is released E Always Use Routing Table When this feature is enabled AlwaysUseRouteTable 1 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 users are able to assign a different SIP URI host name for different called and or calling numbers E Assign Profiles to destination address also when a Proxy is used E Alternative Routing When Proxy isn t used an alternative IP destination for telephone number prefixes is available To
295. et on a circuit different from that to which the receiver is connected Consult the dealer or an experienced radio TV technician for help ACTA Customer information 1 This equipment the VoIP Analog Gateway models MP 118 MP 114 and MP 112 complies with Part 68 of the FCC Rules and the requirements adopted by the ACTA On the bottom of the unit of this equipment is a label that contains among other information a product identifier in the format US AC1ITOOBMP11X3AC If requested this number must be provided to the telephone company 2 This equipment is designed to be connected to the telephone network using an RJ 11C connector which is Part 68 compliant The service order code SOC is 9 0Y and the Facility interface code FIC is 02LS2 3 The REN is used to determine the number of devices that may be connected to a telephone line Excessive RENs on a telephone line may result in the devices not ringing in response to an incoming call In most but not all areas the sum of RENs should not exceed five 5 0 To be certain of the number of devices that may be connected to a line as determined by the total RENs contact the local telephone company The REN for this product is part of the product identifier that has the format US AC1ITOOBMP11X3AC The digits represented by 00 are the REN without a decimal point 4 Should the product causes harm to the telephone network the telephone company will notify you in advance that temporary discontinua
296. etMask 255 255 0 0 LocalControlDefaultGW 0 0 0 0 LocalOAMPAddress 10 31 174 50 LocalOAMSubnetMask 255 255 0 0 LocalOAMDefaultGW 0 0 0 0 IP Routing table parameters RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 255 255 0 RoutingTableGatewaysColumn 10 32 0 1 10 31 0 1 RoutingTableInterfacesColumn 1 0 RoutingTableHopsCountColumn 20 20 2 Use the BootP TFTP utility Section C 6 on page 350 to load and burn fb option the firmware version and the ini file you prepared in the previous step to the MediaPack VLANs and multiple IPs support is available only when the firmware is burned to flash 3 Reset the MediaPack after disabling it on the BootP TFTP utility Version 5 0 259 December 2006 7 T E Reader s Notes SIP User s Manual 260 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 Advanced System Capabilities 10 1 Restoring Networking Parameters to their Initial State You can use the Reset button to restore the MediaPack networking parameters to their factory default values described in Table 4 1 and to reset the username and password Note that the MediaPack returns to the software version burned in flash This process also restores the MediaPack parameters to their factory settings Therefore you must load your previously backed up ini file or the default ini file recei
297. eter DefaultReleaseReason default is GWAPP_NO_ROUTE_TO_DESTINATION 3 or by the GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED 79 reason This counter is incremented as a result of calls that fail due to reasons not covered by the other counters The percentage of established calls from attempted calls The average call duration of established calls This counter indicates the number of attempted fax calls This counter indicates the number of successful fax calls SIP User s Manual 190 Document LTRT 65408 SIP User s Manual 5 Web Management 5 7 1 3 Call Routing Status The Call Routing Status screen provides you with information on the current routing method used by the gateway This information includes the IP address and FQDN if used of the Proxy server the gateway currently operates with Figure 5 57 Call Routing Status Screen Calls Routing Status Call Routing Current Method Routing Table Current Proxy Not Used Current Proxy State Table 5 60 Call Routing Status Parameters Parameter Description Current Call Routing Method Proxy Proxy server is used to route calls Routing Table preferred to Proxy The Tel to IP Routing table takes precedence over a Proxy for routing calls PreferRouteTable 1 Routing Table The Tel to IP Routing table is used to route calls Current Proxy Not Used Proxy server isn t defined IP address and FQDN if exists of
298. eters continues on pages 58 to 64 Parameter Description PRACK Mode PRACK mechanism mode for 1xx reliable responses PRACKMode Disable 0 Supported 1 default Required 2 Note 1 The Supported and Required headers contain the 100rel parameter Note 2 MediaPack sends PRACK message if 180 183 response is received with 100rel in the Supported or the Required headers Channel Select Mode Port allocation algorithm for IP to Tel calls ChannelSelectMode You can select one of the following methods By Dest Phone Number 0 Select the gateway port according to the called number called number is defined in the Endpoint Phone Number table Cyclic Ascending 1 Select the next available channel in an ascending cycle order Always select the next higher channel number in the hunt group When the gateway reaches the highest channel number in the hunt group it selects the lowest channel number in the hunt group and then starts ascending again Ascending 2 Select the lowest available channel Always start at the lowest channel number in the hunt group and if that channel is not available select the next higher channel Cyclic Descending 3 Select the next available channel in descending cycle order Always select the next lower channel number in the hunt group When the gateway reaches the lowest channel number in the hunt group it selects the highest channel number in the hunt group and then starts
299. eters each determines the validity of the parameters IP address and port number of the corresponding SNMP Manager used to receive SNMP traps Checkbox cleared 0 Disabled default Checkbox selected 1 Enabled Up to five IP addresses of remote hosts that are used as SNMP Managers The device sends SNMP traps to these IP addresses Enter the IP address in dotted format notation for example 108 10 1 255 Up to five parameters used to define the Port numbers of the remote SNMP Managers The device sends SNMP traps to these ports Note The first entry out of the five replaces the obsolete parameter SNMPTrapPort The default SNMP trap port is 162 The valid SNMP trap port range is 100 to 4000 Up to five parameters each determines the activation deactivation of sending traps to the corresponding SNMP Manager Disable 0 Sending is disabled Enable 1 Sending is enabled default Version 5 0 179 December 2006 a WH wi AudioCodes MediaPack 5 6 6 2 Configuring the SNMP Community Strings Use the SNMP Community Strings table 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 Section 14 8 1 on page 313 gt To configure the SNMP Community Strings take these 5 steps 1 Access the Management Settings screen Advanced Configuration men
300. eters by completing the following steps e Open the VLAN Settings screen Advanced Configuration menu gt Network Settings gt VLAN Settings option the VLAN Settings screen is displayed e Modify the VLAN parameters to correspond to the values shown in Figure 9 2 below Figure 9 2 Example of the VLAN Settings Screen Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID e Click the Submit button to save your changes Configure the multiple IP parameters by completing the following steps e Open the IP Settings screen Advanced Configuration menu gt Network Settings gt IP Settings option the IP Settings screen is displayed e Modify the IP parameters to correspond to the values shown in Figure 9 3 Note that the OAM Control and Media Network Settings parameters appear only after you select the options Multiple IP Networks or Dual IP in the field IP Networking Mode Note Configure the OAM parameters only if the OAM networking parameters are different from the networking parameters used in the Single IP Network mode Version 5 0 257 December 2006 Ta Ca AudioCodes MediaPack Figure 9 3 Example of the IP Settings Screen IP Settings IP Networking Mode Multiple IP Networks OAM Network Settings IP Address f 0 31 174 50 Subnet Mask 255 255 0 0 Default Gateway Address Control Network Settings
301. f a 19 inch rack unit 1 U high Lifeline provides a wired phone connection to PSTN line when there is no power or the network fails combined FXS FXO gateways provide a Lifeline connection available on all FXS ports LEDs on the front panel that provide information on the operating status of the media gateway and the network interface Reset button on the rear panel that restarts the MP 11x gateway and is also used to restore the MP 11x parameters to their factory default values Version 5 0 21 December 2006 a WH wt AudioCodes MediaPack 1 3 3 MP 124 Hardware Features MP 124 19 inch 1U rugged enclosure provides up to 24 analog FXS ports using a single 50 pin Telco connector LEDs on the front and rear panels that provide information on the operating status of the media gateway and the network interface Reset button on the front panel that restarts the MP 124 gateway and is also used to restore the MP 124 parameters to their factory default values 1 3 4 SIP Features The MediaPack SIP gateway complies with the IETF RFC 3261 standard Reliable User Datagram Protocol UDP transport with retransmissions Transmission Control Protocol TCP Transport layer SIPS using TLS T 38 real time Fax using SIP Note If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message the gateway returns the same rate in the response SDP Works with Proxy or without Proxy using a
302. f time are ignored Note It s recommended to reduce the detection time by 50 msec from the desired value e g if you set the value to 200 msec then enter 150 msec i e 200 minus 50 Max Flash Hook Detection Period Defines the flash hook period in msec for both analog and IP sides msec For the analog side it defines FlashHookPeriod The maximal hook flash detection period for FXS gateways A longer signal is considered offhook onhook event The hook flash generation period for FXO gateways For the IP side it defines the flash hook period that is reported to IP The valid range is 25 to 1500 The default value is 700 msec Note For FXO gateways a constant of 90 msec must be added to the required hook flash period For example to generate a 450 msec hook flash set FlashHookPeriod to 540 SIP User s Manual 160 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 2 5 Configuring the General Media Settings gt To configure the General Media Settings parameters take these 4 steps 1 Open the General Media Settings screen Advanced Configuration menu gt Media Settings gt General Media Settings option the General Media Settings screen is displayed Figure 5 44 General Media Settings Screen DSP Version Template Number 0 Max Echo Canceller Length Default Enable Continuity Tones Disable 2 Configure the Gene
303. field Source number prefix matches the prefix defined in the Source Prefix field Source IP address matches the IP address defined in the Source IP field if applicable Note that number manipulation can be performed using a combination of each of the above criteria or using each criterion independently Note For available notations that represent multiple numbers refer to Section 5 5 3 1 on page 95 Number of Stripped Digits Enter the number of digits that you want 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 Enter the number of digits in brackets that you want to remove from the right of the telephone number prefix Note A combination of the two options is allowed e g 2 3 Prefix Suffix to Add Prefix Enter the number string you want to add to the front of the phone number For example if you enter 9 and the phone number is 1234 the new number is 91234 a Suffix Enter the number string in brackets you want to add to the end of the phone number For example if you enter 00 and the phone number is 1234 the new number is 123400 Note You can enter a prefix and a suffix in the same field e g 9 00 Number of Digits to Leave Enter the number of digits that you want to leave from the right Note The manipulation rules are executed in the following order Num of stripped digits Number of digi
304. fo1 varbind has the MAC address of the gateway Sent when the NAT is placed in front of a gateway that is identified as a symmetric NAT It is cleared when a non symmetric NAT or no NAT replaces the symmetric one This trap is used to indicate the status of the Built In Test BIT The information in the trap contains board hardware elements being tested and their status The information is presented in the additional info fields Sent when an Ethernet link s is down Sent every time the threshold of a Performance Monitored object is crossed The severity field is indeterminate when the crossing is above the threshold and cleared when it goes back under the threshold The source varbind in the trap indicates the object for which the threshold is being crossed Sent at the success or failure of HTTP download SIP User s Manual 312 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 8 14 8 1 In addition to the listed traps the device also supports the following standard traps E coldStart authenticationFailure linkDown linkup entConfigChange SNMP Interface Details This section describes details of the SNMP interface that is required when developing an Element Manager EM for any of the TrunkPack VoP Series products or to manage a device with a MIB browser The gateway offers the following SNMP security features E SNMPv2c community strings E SNMPv3 User based Security Model
305. for Secure Internet Protocol IPsec Policy Enable flag ENABLEMEDIASECURITY Enables or disables Media Security protocol SRTP Link SecuritySettings IKEPOLICYLIFEINSEC This parameter is used for Internet Key Encryption IKE Policy IKE SA LifeTime in seconds Link AKET able Each searched result displays the following e Parameter name hyperlinked to its location in the Embedded Web Server e Brief description of the parameter e Hyperlink in green displaying the URL path to its location in the Embedded Web Server location SIP User s Manual 54 Document LTRT 65408 SIP User s Manual 5 Web Management 3 In the searched result list click the required parameter to open the screen in which the parameter appears The relevant parameter is highlighted in green in the screen for easy viewing Figure 5 4 Searched Parameter Highlighted in Screen IP Networking Mode Single IP Network IP Address 10 8 219 118 Subnet Mask 255 255 0 0 Default Gateway Address 10 8 0 1 DNS Settings DNS Primary Server IP 10 1 1 11 DHCP Settings T Enable DHCP Disable v NAT Settings NAT IP Address 0 0 0 0 Differential Services Network QoS 46 Media Premium QoS 46 Control Premium Qos 46 Gold QoS 26 Bronze QoS 10 Note If the searched parameter is not located the No Matches Found For This String
306. fore a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The SIP gateway generates this response if the called party is off hook and the call cannot be presented as a call waiting call On receiving this response the gateway notifies the User and generates a busy tone This response indicates that the initial request is terminated with a BYE or CANCEL request The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call SIP User s Manual 346 Document LTRT 65408 SIP User s Manual B SIP Compliance Tables B 5 5 5xx Response Server Failure Responses 5xx Response 500 501 502 503 504 505 Internal Server Error Not Implemented Bad gateway Service Unavailable Gateway Timeout Version Not Supported Table B 9 5xx SIP Responses Comments On reception of any of these Responses the GW releases the call sending appropriate release cause to PSTN side The GW generates 5xx response according to PSTN release cause coming from PSTN B 5 6 6xx Response Global Responses 6xx Response Table B 10 6xx SIP Responses Comments 600 Busy Everywhere 603 Decline On reception of any of these Res
307. g follow the procedure to change the trap community string see below Follow the procedure above to delete a read write community name in the row for v2admin The following procedure assumes that a row already exists in the ssCommunityTable for the new trap community string The trap community string can be part of the TrapGroup ReadGroup or ReadWriteGroup If the trap community string is used solely for sending traps recommended it should be made part of the TrapGroup gt 1 To change the trap community string take these 2 steps Add a row to the vacmSecurityToGroupTable with these values SecurityModel 2 SecurityName the new trap community string GroupName TrapGroup ReadGroup or ReadWriteGroup The SecurityModel and SecurityName objects are row indices Modify the SecurityName field in the sole row of the snmpTargetParamsTable Note You must add GroupName and RowStatus on the same set SIP User s Manual 314 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 8 2 SNMP v3 USM Users You can define up to 10 User based Security Model USM users USM users are referred to as v3 users Each v3 user can be associated with an authentication type none MD5 or SHA 1 and a privacy type none DES 3DES or AES Table 14 3 SNMP v3 Security Levels Security Level noAuthNoPriv 1 authNoPriv 2 authPriv 3 Authentication None MD5 or SHA 1 MD5 or SHA 1 Privacy None None
308. g Lines Ethernet RS 232 AC power supply socket Front Panel Reset Button Physical Enclosure Dimensions 32 tones single tone dual tones or AM tones programmable frequency amp amplitude 64 frequencies in the range 300 to 1980 Hz 1 to 4 cadences per tone up to 4 sets of ON OFF periods 32 dB to 31 dB in steps of 1 dB 32 dB to 31 dB in steps of 1 dB Group 3 fax relay up to 14 4 kbps with auto fallback T 38 compliant real time fax relay Tolerant network delay up to 9 seconds round trip Auto switch to PCM or ADPCM on V 34 or V 90 modem detection SIP RFC 3261 RTP RTCP packetization IP stack UDP TCP RTP Remote Software load TFTP HTTP and HTTPS Loop start Motorola PowerQUICC 860 SDRAM 64 MB AudioCodes AC482 VoIP DSP 24 Analog FXS phone or fax ports loop start RJ 11 10 100 Base TX RS 232 Terminal Interface DB 9 Channel status and activity LEDs 50 pin Telco shielded connector 10 100 Base TX RJ 45 shielded connector Console port DB 9 100 240 0 8A max Resets the MP 124 1U 19 inch Rack Width 445mm 17 5 in Height 44 5mm 1 75 in Depth 269mm 10 6 in Weight 1 8 kg 4 Ib Environmental Operational 5 to 40 C 41 to 104 F Storage 25 to 70 C 77 to 158 F Humidity 10 to 90 non condensing Mounting Rack mount Desktop Electrical 100 240 VAC Nominal 50 60 Hz Version 5 0 337 December 2006 Ta WH wt AudioCodes MediaPack Table 16 2 MP 124 Func
309. g Mode to Two Stage IsTwoStageDial 1 Version 5 0 245 December 2006 7 T E Reader s Notes SIP User s Manual 246 Document LT RT 65408 SIP User s Manual 9 Networking Capabilities 9 9 1 9 2 Networking Capabilities Ethernet Interface Configuration Using the parameter EthernetPhyConfiguration users can control the Ethernet connection mode Either the manual modes 10 Base T Half Duplex 10 Base T Full Duplex 100 Base TX Half Duplex 100 Base TX Full Duplex or Auto Negotiate mode can be used Auto Negotiation falls back to Half Duplex mode when the opposite port is not Auto Negotiate but the speed 10 Base T 100 Base TX in this mode is always configured correctly Note that configuring the gateway to Auto Negotiate mode while the opposite port is set manually to Full Duplex either 10 Base T or 100 Base TX is invalid as it causes the gateway to fall back to Half Duplex mode while the opposite port is Full Duplex It is also invalid to set the gateway to one of the manual modes while the opposite port is either Auto Negotiate or not exactly matching both in speed and in duplex mode Users are encouraged to always prefer Full Duplex connections to Half Duplex ones and 100 Base TX to 10 Base T due to the larger bandwidth It is strongly recommended to use the same mode in both link partners Any mismatch configuration can yield unexpected functioning of the Ethernet connection
310. gList MGR Params v2cparams where N is an unused number between 0 and 4 2 Add a row to the tgtAddressMaskTable table with these values Name mgrN tgtAddressMask 255 255 255 255 0 An alternative to the above procedure is to set the tgtAddressMask column while you are creating other rows in the table The following procedure assumes that there is at least one configured read write community There are currently two or more Trusted Managers The taglist for columns for all rows in the srCommunityTable are currently set to MGR This procedure must be performed from one of the existing trusted managers but not the one that is being deleted gt To delete a Trusted Manager not the final one take this step m Remove the appropriate row from the snmpTargetAddrTable The change takes effect immediately The deleted trusted manager cannot access the device The agent automatically removes the row in the tgtAddressMaskTable SIP User s Manual 318 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 8 4 14 8 5 14 8 5 1 The following procedure assumes that there is at least one configured read write community There is currently only one Trusted Manager The taglist for columns for all rows in the srCommunityTable are currently set to MGR This procedure must be performed from the final Trusted Manager gt To delete the final Trusted Manager take these 2 steps 1 Set the value of the TransportLabel field o
311. gateway 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 SIP User s Manual 72 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 5 Supported Coders and their Attributes ogee Payload Silence Coder Name Packetization Time Rate Type Suppression G 711Alaw 10 20 default 30 40 50 Disable 0 g711Alaw64k 60 80 100 120 Always 64 Always amp Enable 1 G 711 law 10 20 default 30 40 50 Disable 0 g711Ulaw64k 60 80 100 120 Aways 64 Aways 0 Enable 1 Disable 0 G 729 10 20 default 30 40 50 Enable 1 9729 60 Aways g Always 18 Enable w o Adaptations 2 G 723 1 5 3 0 6 3 1 Disable 0 97231 30 default 60 90 default Always 4 Enable 1 G 726 10 20 default 30 40 50 Bay o Dynamic 0 Disable 0 9726 60 80 100 120 40 3 120 Enable 1 T 38 N A N A N A N A t38fax G 711A law_VBD 10 20 default 30 40 50 60 Always 64 Dynamic 0 N A 9711AlawVbd 80 100 120 120 G 711U law_VBD 10 20 default 30 40 50 60 Always 64 Dynamic 0 N A 9711UlawVbd 80 100 120 120 Transparent N A Table 5 6 ini File Coder Parameter Parameter Description CoderName Defines the gateway s coder list up to five coders can be configur
312. gateways E The embedded Command Line Interface CLI accessed via Telnet or RS 232 refer to Section 4 2 4 on page 44 m DHCP refer to Section 7 2 on page 212 m Use the Reset button at any time to restore the MediaPack networking parameters to their factory default values refer to Section 10 1 on page 261 Version 5 0 41 December 2006 7a Ta r wt AudioCodes MediaPack 4 2 1 Assigning an IP Address Using HTTP gt 1 To assign an IP address using HTTP take these 8 steps Disconnect the MediaPack from the network and reconnect it to your PC using one of the following two methods e Use a standard Ethernet cable to connect the network interface on your PC to a port on a network hub switch Use a second standard Ethernet cable to connect the MediaPack to another port on the same network hub switch e Use an Ethernet cross over cable to directly connect the network interface on your PC to the MediaPack Change your PC s IP address and subnet mask to correspond with the MediaPack factory default IP address and subnet mask shown in Table 4 1 For details on changing the IP address and subnet mask of your PC refer to Windows Online Help Start gt Help Access the MediaPack Embedded Web Server refer to Section 5 3 on page 51 In the Quick Setup screen shown in Figure 4 1 set the MediaPack IP Address Subnet Mask and Default Gateway IP Address fields under IP Configur
313. ge is 1 to 10 seconds The default value is 4 seconds SIP User s Manual 74 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 7 DTMF amp Dialing Parameters continues on pages 74 to 76 Parameter Declare RFC 2833 in SDP RxDTMFOption 1 to 5 Tx DTMF Option TxDTMFOption RFC 2833 Payload Type RFC2833PayloadType Hook flash Option Description Defines the supported Receive DTMF negotiation method No 0 Don t declare RFC 2833 telephony event parameter in SDP Yes 8 Declare RFC 2833 telephony event parameter in SDP default The MediaPack is designed to always be receptive to RFC 2833 DTMF relay packets Therefore it is always correct to include the telephony event parameter as a default in the SDP However some gateways use the absence of the telephony event from the SDP to decide to send DTMF digits in band using G 711 coder if this is the case you can set RxDTMFOption 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 RFC2833PayloadType default 1 INFO Nortel Sends DTMF digits according to IETF lt draft choudhuri sip info digit O00 gt 2 NOTIFY Sends DTMF digits according to lt draft mahy sipping signaled digits 01 gt 3 INFO Cisco Sends DTMF digits according to Cisco format 4 RFC 2833 No
314. ges For this reason no assumption is made on the contents of the messages other than the minimum requirements of its priority Syslog uses UDP as its underlying transport layer mechanism The UDP port can be defined using SyslogServerPort parameter default port is 514 The Syslog message is transmitted as an ASCII American Standard Code for Information Interchange message The message starts with a leading lt less than character followed by a number which is followed by a gt greater than character This is optionally followed by a single ASCII space The number described above is known as the Priority and represents both the Facility and Severity as described below The Priority number consists of one two or three decimal integers For example lt 37 gt Oct 11 16 00 15 mymachine su su root failed for lonvick on dev pts 8 13 3 1 Syslog Servers Users can use the provided AudioCodes Syslog server ACSyslog or any other third party Syslog servers Examples of Syslog servers available as shareware on the Internet E Kiwi Enterprises www kiwisyslog com m The US CMS Server uscms fnal gov hanlon uscms_server m TriAction Software www triaction nl Products SyslogDaemon asp E Netal SL4NT 2 1 Syslog Daemon www netal com A typical Syslog server application enables filtering of the messages according to priority IP sender address time date etc Version 5 0 303 December 2006 Ta
315. grading the MediaPack loading new software onto the gateway using the BootP TFTP configuration utility m From version 4 4 to version 4 4 or to any higher version the device retains its configuration ini file However the auxiliary files CPT logo etc may be erased E From version 4 6 to version 4 6 or to any higher version the device retains its configuration ini file and auxiliary files CPT logo etc You can also use the Software Upgrade wizard available through the Web Interface refer to Section 5 9 1 on page 197 To save the cmp file to non volatile memory use the fb command line switches If the file is not saved the gateway reverts to the old version of software after the next reset For information on using command line switches refer to Section C 11 6 on page 359 Version 5 0 213 December 2006 Ta WH wt AudioCodes MediaPack 7 3 2 Vendor Specific Information Field The MediaPack uses the vendor specific information field in the BootP request to provide device related initial startup information The BootP TFTP configuration utility displays this information in the Client Info column refer to Figure C 1 Note This option is not available on DHCP servers The Vendor Specific Information field is disabled by default To enable disable this feature set the ini file parameter ExtBootPReqEnable Table 5 55 on page 182 or use the be command line switch refer to Tab
316. gt Tel and Tel gt IP routing tables and their associated parameters 5 5 5 1 General Parameters Use this screen to configure the gateway s IP gt Tel and Tel gt IP routing parameters gt To configure the general parameters under Routing Tables take these 4 steps 1 Open the General Parameters screen Protocol Management menu gt Routing Tables submenu gt General option the General Parameters screen is displayed Figure 5 16 Routing Tables General Parameters Screen General Parameters Add Hunt Group ID as Prefix No Add Port Number as Prefix No IP to Tel Remove Routing Table Prefix No Enable Alt Routing Tel to IP Disable Alt Routing Tel to IP Mode None Max Allowed Packet Loss for Alt Routing 20 Max Allowed Delay for Alt Routing msec 250 2 Configure the general parameters under Routing Tables according to Table 5 16 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 16 Routing Tables General Parameters continues on pages 98 to 99 Parameter Add Hunt Group ID as Prefix AddTrunkGroupAsPrefix Add Port Number as Prefix AddPortAsPrefix Description No 0 Don t add hunt group ID as prefix default Yes 1 Add hunt group ID as prefix to called number If enabled then the hunt group ID is added as a prefix to the destina
317. gure 5 61 MediaPack FXS Channel Status Screen Channel Status Ad 2 ee Co Color Code Ke Q Inactive J Handset Offhook x RTP Active The color of each channel shows the call status of that channel Refer to Table 5 62 below for information on the different statuses a call can have Table 5 62 Channel Status Color Indicators Indicator Label Description Inactive Indicates this channel is currently onhook RTP Active Indicates an active RTP stream Not Connected FXO only Indicates that no analog line is connected to this port Handset Offhook Indicates this channel is offhook but there is no active RTP session Version 5 0 195 December 2006 Ta r AudioCodes MediaPack gt To monitor the details of a specific channel take these 3 steps 1 Click the numbered port icon of the specific channel whose detailed status you need to check monitor a shortcut menu appears 2 From the shortcut menu choose Port Settings the channel specific Channel Status screen appears shown in Figure 5 62 3 Click the submenu links to check view a specific channel s parameter settings Figure 5 62 Channel Status Details Screen SIP Channel Status Static Information Endpoint Status ACTIVE Assigned Phone Number 100 Hunt Group default 0 MWI Information Associated Calls Information Call ID 265821508dMlu 10 8 58 1 Call Originator TEL Source Tel Number 100
318. haracters SNMPTrapCommunitySiring The default string is trapuser SIP User s Manual 180 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 6 3 Configuring SNMP V3 Use the SNMP V3 Table to configure authentication and privacy for up to 10 SNMP V3 users For detailed information on SNMP community strings refer to Section 14 8 1 on page 313 gt To configure the SNMP V3 Users take these 5 steps 1 Access the Management Settings screen Advanced Configuration menu gt Management Settings the Management Settings screen is displayed Figure 5 51 2 Open the SNMP V3 Setting screen by clicking the arrow sign gt to the right of the SNMP V3 Table label the SNMP V3 Setting screen is displayed Figure 5 54 3 Configure the SNMP V3 Setting parameters according to Table 5 54 below Click the Apply Row Settings button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 gt Figure 5 54 SNMP V3 Setting Screen SNMP V3 Setting 0 0 O o lo H t li To delete an SNMP V3 user select the Index radio button corresponding to the SNMP V3 user row entry to which you want to delete and then click the Delete Row button Table 5 54 SNMP V3 Setting Parameters Parameter Description Index The table index Row number Username Name of the SNMP v3 user This name must be unique SNMPUsers_ Username Au
319. he Caller Display Information screen Protocol Management menu gt Endpoint Settings submenu gt Caller ID option the Caller Display Information screen is displayed Figure 5 29 Caller Display Information Screen Caller ID Name Presentation Susan C Allowed Port 2 FXS Lee Y Allowed v Port3 FXS Mike D Restricted v Pot 4 FXS Private IE Restricted z Pot 5 FXO Allowed yj Port 6 FXO Allowed x Port7 FXO Allowed v Allowed In the Caller ID Name field enter the Caller ID string The Caller ID string can contain up to 18 characters Note that when the FXS gateway receives Private or Anonymous strings in the From header it doesn t send the calling name or number to the Caller ID display Version 5 0 121 December 2006 a WH wt AudioCodes MediaPack 3 From the Presentation drop down list select e Allowed 0 to send the string in the Caller ID Name field when a Tel gt IP call is made using this VoIP gateway port e Restricted 1 if you don t want to send this string When the Presentation field is set to Restricted the caller identity is passed to the remote side using only the P Asserted Identity and P Preferred Identity headers AssertedidMode The value of the Presentation field can optionally be ove
320. herefore this method should only be used as a backup mode Relevant parameters EnableSilenceDisconnect and FarEndDisconnectSilencePeriod with DSP templates number 2 or 3 A special DTMF code A digit pattern that when received from the Tel side indicates to the gateway to disconnect the call Relevant ini file parameter TelDisconnectCode Version 5 0 225 December 2006 Ta WH wt AudioCodes MediaPack 8 4 1 4 8 4 2 E Interruption of RTP stream Relevant parameters BrokenConnectionEventTimeout and DisconnectOnBrokenConnection Note This method operates correctly only if silence suppression is not used Protocol based termination of the call from the IP side Note The implemented disconnect method must be supported by the CO or PBX DID Wink The gateway 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 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 fo
321. iaPack gateway the call s destination number is compared to the list of prefixes defined in the Tel to IP Routing table described in Section 5 5 5 2 on page 100 The Tel to IP Routing table is scanned for the destination number s prefix starting at the top of the table When an appropriate entry destination number matches one of the prefixes is found the prefix s corresponding destination IP address is checked If the destination IP address is disallowed an alternative route is searched for in the following table entries Destination IP address is disallowed if no ping to the destination is available ping is continuously initiated every 7 seconds when an inappropriate level of QoS was detected or when 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 The MediaPack gateway matches the rules starting at the top of the table For this reason enter the main IP route above any alternative route 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 configurable methods are applied Connectivity The destination IP address is queried periodically currently only by ping E QoS The QoS of an IP connection is determined according to RTCP statistics of p
322. iaSecurity to 1 described in Table 5 50 When SRTP is used the channel capacity is reduced refer to the parameter EnableMediaSecurity The gateway only supports the AES 128 in CM mode cipher suite Figure 12 11 Example of crypto Attributes Usage a crypto 1 AES_CM_128_HMAC_SHA1_80 inline PSKoMpH1Cg b5X0YLuSvNriImEh dAe a crypto 2 AES_CM_128_HMAC_SHA1_32 inline IsPtLoGkBf9a c6XVzRuMqH1DnEiAd Version 5 0 293 December 2006 Ta WH wt AudioCodes MediaPack 12 4 12 4 1 RADIUS Login Authentication Users can enhance the security and capabilities of logging to the gateway s Web and Telnet embedded servers by using a Remote Authentication Dial In User Service RADIUS to store numerous usernames passwords and access level attributes Web only allowing multiple user management on a centralized platform RADIUS RFC 2865 is a standard authentication protocol that defines a method for contacting a predefined server and verifying a given name and password pair against a remote database in a secure manner When accessing the Web and Telnet servers users must provide a valid username and password When RADIUS authentication isn t used the username and password are authenticated with the Embedded Web Server s usernames and passwords of the primary or secondary accounts refer to Section 5 2 1 on page 49 or with the Telnet server s username and password stored internally in the gateway s memory When RADI
323. if RingsBeforeCallerlD 0 FXO Gateway FXO detects rings on line FXO seizes line off hook only after receiving 200 OK even after receiving 183 to enable routing to voice mail on the PBX side Version 5 0 227 December 2006 7a Ta P wt AudioCodes MediaPack 8 4 2 2 Collecting Digits Mode When automatic dialing is not defined the gateway collects the digits The SIP call flow diagram below illustrates the Collecting Digits Mode Figure 8 4 Call Flow for Collecting Digits Mode SIP Client ae 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 8 4 2 3 Ring Detection Timeout The ini file parameters IsWaitForDialTone and WaitForDialTone apply to Ring Detection Timeout The operation of Ring Detection Timeout depends on the following H No automatic dialing and Caller ID is enabled if the second ring signal doesn t arrive for Ring Detection Timeout the gateway 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 Ring Detection Timeout the FXO releases the IP call Ring Detection Timeout supports full ring cycle of ring on and ring off from ring start to ring start SIP User s Manual 228 Document LTRT 65408 SIP User s
324. ight 00 It s mandatory that the last time period per rule ends at midnight 00 This prevents undefined time frames in a day The gateway selects the time period by comparing the gateway s current time to the end time of each time period of the selected Charge Code The gateway 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 Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 11 Charge Codes Table ini File Parameter Parameter Name in ini File Parameter Format ChargeCode ChargeCode_ lt Charge Code ID gt lt 1 period end time gt lt 1 period pulse interval gt lt 1 period pulses on answer gt lt 2 period end time gt lt 2 period pulse interval gt lt 2 period pulses on answer gt lt 3 period end time gt lt 3 period pulse interval gt lt 3 period pulses on answer gt lt 4 period end time gt lt 4 period pulse interval gt lt 4 period pulses on answer gt For example ChargeCode_1 07 30 1 14 20 2 20 15 1 00 60 1 ChargeCode_2 05 60 1 14 20 1 00 60 1 ChargeCode_3 00 60 1 Note Up to 25 different metering rules can be defined by repeating the parameter 25 times SIP User s Manual 88 Docum
325. ile 3 During the installing you are prompt to determine the Destination Folder under which the package is installed it is advised to provide a non spaced path such as c directory_name_without_spaces 4 In the following screen configuration uncheck the Build html docs and Configure CPAN pm checkboxes if they are present If you are prompted to bring nmake answer no 5 After the installation is complete add the path perl bin and path apache2 bin path stands for the path that was previously specified in the Destination Folder directories to the system known path Open the Control Panel gt System gt Advanced gt Environment Variables inside the System Variables dialog box choose Path and click the Edit button in the opened Variable Value checkbox append both of the paths to the existing list Restart window in order to activate the new paths 6 Open the Apache2 conf httpd conf file for editing and set the parameter MaxKeepAliveRequests to 0 enables an unlimited number of requests during a persistent connection required for multiple consecutive HTTP PUT requests for uploading the file Version 5 0 383 December 2006 Ta WH wt AudioCodes MediaPack 7 Open the Apache2 conf perl conf file for editing and add the line Script PUT perl put cgi after the last line in the following section note that if the following section is omitted or different in the file insert
326. ile to the gateway the recommended policy is 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 includes a Format line that defines the columns of the table to be modified this may vary from ini file to ini file for the same table The Format line must only include columns that can be modified parameters that are not specified as Read Only An exception is Index fields that are always mandatory In the example provided in Table 10 1 all fields except for the Time Connected field are loaded Structure of Parameter Tables in the ini File Tables are composed of four elements E Title of the table The name of the table in square brackets e g MY_TABLE_NAME E A Format line Specifies the columns of the table by their string names that are to be configured The first word of the Format line must be FORMAT followed by the names of the Indices fields and an equal sign After the equal sign the names of the columns are listed Items must be separated by a comma The Format line must end with a semicolon Data line s Contain the actual values of the parameters The values are interpreted according to the Format line The first word of the Data line must be the table s string name followed by the Index fields Items must be separated by a comma e A Data line must end with
327. ime Before Waiting Indication 0 Waiting Beep Duration 300 Enable Caller ID Disable Caller ID Type Bellcore Hook Flash Code MWI Parameters Enable Mv Disable MWI Analog Lamp Disable MWI Display Disable Subscribe to MV No MWI Server IP Address MWI Subscribe Expiration Time MWI Subscribe Retry Time Stutter Tone Duration Conference Enable 3 Way Conference Disable Establish Conference Code Conference ID conf Version 5 0 83 December 2006 Ta fal AudioCodes MediaPack 2 Configure the supplementary services parameters according to Table 5 9 3 Click the Submit button to save your changes or click the Subscribe for MWI or Un Subscribe for MWI buttons to save your changes and to subscribe unsubscribe to the MWI server 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 9 Supplementary Services Parameters continues on pages 84 to 86 Parameter Enable Hold EnableHold Hold Format HoldFormat Enable Transfer EnableTransfer Transfer Prefix xferPrefix Enable Call Forward EnableForward Enable Call Waiting EnableCallWaiting Number of Call Waiting Indications NumberOfWaitingIndicati ons Description No 0 Disable the Hold service Yes 1 Enable the Hold service default
328. in Table 5 55 on page 182 Version 5 0 35 December 2006 ful AudioCodes MediaPack 3 2 Installing the MP 124 3 2 1 3 2 2 3 2 3 3 2 3 1 To install the MP 124 take these 4 steps Unpack the MP 124 refer to Section 3 2 1 below Check the package contents refer to Section 3 2 2 below Mount the MP 124 refer to Section 3 2 3 on page 36 Cable the MP 124 refer to Section 3 2 4 on page 38 After connecting the MP 124 to the power source the Ready and LAN LEDs on the front panel turn to green after a self testing period of about 1 minute Any malfunction changes the Ready LED to red When you have completed the above relevant sections you are then ready to start configuring the gateway Chapter 4 on page 41 Pen Unpacking To unpack the MP 124 take these 6 steps Open the carton and remove packing materials Remove the MP 124 gateway from the carton Check that there is no equipment damage Check retain and process any documents Notify AudioCodes or your local supplier of any damage or discrepancies Retain any diskettes or CDs CM ee PS Yy Package Contents Ensure that in addition to the MP 124 the package contains E AC power cable m 2 short equal length brackets and bracket to device screws for the 19 inch rack installation A CD with software and documentation may be included m The MediaPack Fast Track Installation Guide Mounting the MP 124 The MP 124 can be moun
329. information isn t sent to display default Enable 1 MWI information is sent to display If enabled the gateway generates an MWI FSK message that is displayed on the MWI display This parameter is applicable only to FXS gateways Version 5 0 85 December 2006 Ta WH wt AudioCodes MediaPack Table 5 9 Supplementary Services Parameters continues on pages 84 to 86 Parameter Subscribe to MWI EnableMWISubscription MWI Server IP Address MWIServerIP MWI Subscribe Expiration Time MWIExpirationTime MWI Subscribe Retry Time SubscribeRetryTime Stutter Tone Duration StutterToneDuration Conference Parameters Enable 3 Way Conference Enable3WayConference Establish Conference Code ConferenceCode Conference ID ConferencelD Description Disable 0 Disable MWI subscription default Enable 1 Enable subscription to MWI to MWIServerIP address Note Use the parameter SubscriptionMode described in Table 5 35 on page 132 to determine whether the gateway subscribes separately per endpoint of for the entire gateway MWI server IP address If provided the gateway subscribes to this IP address Can be configured as a numerical IP address or as a domain name If not configured the Proxy IP address is used instead MWI subscription expiration time in seconds The default is 7200 seconds The range is 10 to 72000 Subscription retry time in seconds The default is 120 se
330. ing Device Software Update Access to Restricted Domains Non Authorized Access 00000000 Sensitive Parameters Value Change 2 Configure the Management Settings according to Table 5 51 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 SIP User s Manual 176 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 51 Management Settings Parameters continues on pages 177 to 178 Parameter Description Syslog Settings Syslog Server IP address IP address in dotted format notation of the computer you are using to run SyslogServerIP the Syslog Server The Syslog Server is an application designed to collect the logs and error messages generated by the VoIP gateway For information on the Syslog refer to Section 13 2 on page 301 Syslog Server Port Defines the UDP port of the Syslog server SyslogServerPort The default value i e port is 514 Enable Syslog Enable 1 Send the logs and error message generated by the gateway to EnableSyslog the Syslog Server If you select Enable you must enter an IP address in the Syslog Server IP address field Disable 0 Logs and errors are not sent to the Syslog Server default Note 1 Syslog messages may increase the network traffic Note 2 Logs are also sent to the RS 232 serial port for information on establishing a serial communi
331. ing an IP Address Using the CLI First access the CLI using a standard Telnet application or using a serial communication software e g HyperTerminal connected to the MediaPack RS 232 port refer to Section 4 2 4 1 below Then assign the MediaPack an IP address refer to Section 4 2 4 2 below 4 2 4 1 Access the CLI gt To access the CLI via the Embedded Telnet Server take these 3 steps 1 Enable the Embedded Telnet Server a b Access the MediaPack Embedded Web Server refer to Section 5 3 on page 51 Set the parameter Embedded Telnet Server under Advanced Configuration gt Network Settings gt Application Settings to Enable Unsecured or Enable Secured SSL Click the Maintenance button on the main menu bar the Maintenance Actions screen is displayed From the Burn to FLASH drop down list select Yes Click the Reset button the MediaPack is shut down and re activated A message about the waiting period is displayed The screen is refreshed 2 Use a standard Telnet application to connect to the MediaPack Embedded Telnet Server Note that if the Telnet server is set to SSL mode a special Telnet client is required on your PC to connect to the Telnet interface over a secured connection 3 Login using the username Admin and password Admin gt To access the CLI via the RS 232 port take these 2 steps 1 Connect the RS 232 port to your PC For the MP 124 refer to
332. ing the establishment of an IP gt Tel call Note Applicable only to FXO gateways for single stage dialing when waiting for dial tone IsWaitForDialTone is disabled 2 The delay between the time when Wink is detected and dialing is begun during the establishment of an IP gt Tel call for DID lines EnableDIDWink 1 3 For call transfer The delay after hook flash is generated and dialing is begun The valid range in milliseconds is 0 to 20000 20 seconds The default value is 1000 1 second Note Applicable only to FXO gateways for Tel IP calls The Ring Detection timeout is used differently for normal and for automatic dialing If automatic dialing is not used and if Caller ID is enabled then the FXO gateway 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 doesn t arrive for Ring Detection Timeout the gateway doesn t initiate a call to IP When automatic dialing is used the FXO gateway initiates a call to IP when 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 ringing signal stops for Ring Detection Timeout the FXO gateway Releases the IP call Usually set to a value between 5 and 8 The default is 8 seconds Reorder Tone Duration sec Busy or Reorder tone duration seconds the FXO gateway plays
333. ing the Start Software Upgrade button that no traffic is running on the device After clicking this button a device reset is mandatory Even if you choose to cancel the process in the middle the device resets itself and the previous configuration burned to flash is reloaded Version 5 0 197 December 2006 Ta r K AudioCodes MediaPack gt To use the Software Upgrade Wizard take these 11 steps Stop all traffic on the MediaPack refer to the note above 2 Open the Software Upgrade Wizard Software Update menu gt Software Upgrade Wizard the Start Software Upgrade screen appears oh Figure 5 63 Start Software Upgrade Screen At this point the process can be canceled with no consequence to the MediaPack click the Cancel button If you continue the process by clicking the Start Software Upgrade button the process must be followed through and completed with a MediaPack reset at the end If you click the Cancel button in any of the subsequent screens the MediaPack is automatically reset with the configuration that was previously burned in flash memory 3 Click the Start Software Upgrade button the Load a cmp file screen appears Figure 5 64 Note When in the Wizard process the rest of the Web application is unavailable and the background Web screen is disabled After the process is completed access to the full Web application is restored SIP User s Manual 198 Document
334. ini files are loaded Note 2 The optional string lt MAC gt is replaced with the gateway s MAC address Therefore the gateway requests an ini file name that contains its MAC address This option enables loading different configurations for specific gateways Note 3 The maximum length of the URL address is 99 characters Specifies the name of the Prerecorded Tones file and the location of the server IP address or FQDN from which it is loaded http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the CPT file and the location of the server IP address or FQDN from which it is loaded http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the FXS coefficients file and the location of the server IP address or FQDN from which it is loaded http server_name file https server_name ffile Note The maximum length of the URL address is 99 characters Specifies the name of the User Information file and the location of the server IP address or FQDN from which it is loaded http server_name file https server_name ffile Note The maximum length of the URL address is 99 characters Enables 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
335. initions of the Call Progress Tones levels and frequencies that are detected generated by the MediaPack The second section contains the characteristics of the distinctive ringing signals that are generated by the MediaPack Users can either use one of the supplied MediaPack configuration daft files or construct their own file To construct their own configuration file users are recommended to modify the supplied usa_tone ini file in any standard text editor to suit their specific requirements and to convert it the modified ini file into binary format using the TrunkPack Downloadable Conversion Utility For the description of the procedure on how to convert CPT ini file to a binary dat file refer to Section E 1 1 on page 364 Note that only the dat file can be loaded to the MediaPack gateway To load the Call Progress Tones dat file to the MediaPack use the Embedded Web Server refer to Section 5 6 4 on page 166 or the ini file refer to Section 5 9 2 1 on page 203 15 1 1 Format of the Call Progress Tones Section in the ini File Users 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 Hz to 1980 Hz or an Amplitude Modulated AM In total 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 Not
336. ion is made For example you can route a call to a specific hunt group according to its original number and then you can remove add a prefix to that number before it is routed To control when number manipulation is done set the IP to Tel Routing Mode described in Table 5 18 and the Tel to IP Routing Mode described in Table 5 17 parameters Possible uses for number manipulation can be as follows E To strip add dialing plan digits from to the number For example a user could dial 9 in front of each number in order to indicate an external line This number 9 can be removed here before the call is setup B Allow disallow Caller ID information to be sent according to destination source prefixes For detailed information on Caller ID refer to Section 5 5 9 3 on page 121 gt To configure the Number Manipulation tables take these 5 steps 1 Open the Number Manipulation screen you want to configure Protocol Management menu gt Manipulation Tables submenu the relevant Manipulation table screen is displayed Figure 5 14 shows the Source Phone Number Manipulation Table for Tel gt IP calls Figure 5 14 Source Phone Number Manipulation Table for Tel gt IP calls Num of Prefix Suffix to Number of Dest Prefix Source Prefix Stripped Digits to Presentation Digits Add Leave Allowed z Resticted z af amos fo Not Configured fsa Not Configurer Not Configured z Not Configured z
337. ion File Screen Configuration File Get the ini file from the device to your computer Send the ii file from your computer to the device Browse _ Send ini File The device will perform a Reset after sending the sn file To back up the ini file take these 4 steps Click the Get ini File button the File Download window opens Click the Save button the Save As window opens Navigate to the folder where you want to save the ini file Click the Save button the VoIP gateway copies the ini file into the folder you selected KURS y To restore the ini file take these 4 steps Click the Browse button Navigate to the folder that contains the ini file you want to load Select the file and then click the Open button the name and path of the file appear in the field beside the Browse button 4 Click the Send ini File button and then click OK at the prompt the gateway is automatically reset from the cmp version stored on the flash memory t E Ae Version 5 0 165 December 2006 Ta A wi AudioCodes MediaPack 5 6 4 Regional Settings The Regional Settings screen enables you to set and view the gateway s internal date and time and to load to the gateway the following configuration files Call Progress Tones coefficient and Voice Prompts currently not applicable to MediaPack gateways For detailed information on the configuration files refer to Chapter 6 on page
338. ion of equipment nameplate ratings should be used when addressing this concern Reliable Earthing Reliable earthing of rack mounted equipment should be maintained Particular attention should be given to supply connections other than direct connections to the branch circuit e g use of power strips 3 2 3 2 Installing the MP 124 in a 19 inch Rack The MP 124 is installed into a standard 19 inch rack by the addition of two short equal length supplied brackets The MP 124 with brackets for rack installation is shown in Figure 3 10 gt To install the MP 124 in a 19 inch rack take these 7 steps Remove the two screws on one side of the device nearest the front panel 1 2 Insert the peg on one of the brackets into the third air vent down on the column of air vents nearest the front panel Swivel the bracket until the holes in the bracket line up with the two empty screw holes on the device Use the supplied screws to attach the bracket to the side of the device Repeat steps 1 to 4 to attach the second bracket to the other side of the device Position the device in the rack and line up the bracket holes with the rack frame holes Use four standard rack screws not supplied to attach the device to the rack 9 oe Figure 3 10 MP 124 with Brackets for Rack Installation Version 5 0 af December 2006 7a VT r ud AudioCodes MediaPack 3 2 4 Cabling the MP 124 Cable your MP 124 according to each section of Ta
339. ion of general attributes of the MIB and specific attributes of the monitored objects Data sub tree The monitoring results are presented in tables Each table includes one or two indices When there are two indices the first index is a sub set in the table e g trunk number and the second or a single where there is only one index represents the interval number present 0 previous 1 and the one before 2 SIP User s Manual 308 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 5 14 6 The MIBs include e acPMMedia for media voice related monitoring e g RTP DSP s e acPMControl for Control Protocol related monitoring e g connections commands e acPMAnalog for analog channels in offhook state e acPMSystem for general system related monitoring The log trap acPerformanceMonitoringThresholdCrossing non alarm is sent out every time the threshold of a Performance Monitored object is crossed The severity field is indeterminate when the crossing is above the threshold and cleared when it falls bellow the threshold The source varbind in the trap indicates the object for which the threshold is being crossed Total Counters The counter s attribute total accumulates counter values since the board s most recent restart The user can reset the total s value by setting the Reset Total object Each MIB module has its own Reset Total object as follows
340. ion per device Each device loads a file named after its MAC address e g config_00908F033512 ini IniFileURL http www corp com config_ lt MAC gt ini Reset the device after configuration is updated The device resets after all of the files are processed ResetNow 1 You can modify the master_configuration ini file or any of the config_ lt MAC gt ini files at any time The MediaPack queries for the latest version every 60 minutes and applies the new settings immediately 4 For additional security use HTTPS or FTPS The MediaPack supports HTTPS RFC 2818 and FTPS using the AUTH TLS method lt draft murray auth ftp ssl 16 gt for the Automatic Update mechanism 5 To load configuration files from an NFS server the NFS file system parameters should be defined in the configuration ini file The following is an example of an ini file for loading files from NFS servers using NFS version 2 Define NFS servers for Automatic Update NFSServers FORMAT NFSServers_Index NFSServers_HostOrIP NFSServers_RootPath NFSServers_NfsVersion NFSServers 1 10 31 2 10 usr share 2 NFSServers 2 192 168 100 7 d shared 2 NFSServers CetFileUrl file 10 31 2 10 usr share public usa_tones dat VpFileUrl file 192 168 100 7 d shared audiocodes voiceprompt dat SIP User s Manual 264 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 4 Startup Proces
341. ion with leading PBXs IP PBXs Softswitches and SIP servers Spans a range of 2 to 24 FXS FXO analog ports Selectable G 711 or multiple Low Bit Rate LBR coders per channel T 38 fax with superior performance handling a round trip delay of up to nine seconds Echo Canceler Jitter Buffer Voice Activity Detection VAD and Comfort Noise Generation CNG support Comprehensive support for supplementary services Web Management for easy configuration and installation EMS for comprehensive management operations FCAPS Simple Network Management Protocol SNMP and Syslog support SMDI support for Voice Mail applications ThroughPacket proprietary feature that aggregates payloads from several channels into a single IP packet to reduce bandwidth overhead Supports load balancing with proxy T 38 fax fallback to PCM or NSE Can be integrated into a Multiple IPs and a VLAN aware environment Capable of automatically updating its firmware version and configuration Secured Web access HTTPS and Telnet access using SSL TLS IPSec and IKE protocols are used in conjunction to provide security for control e g SIP and management e g SNMP and Web protocols Secured RTP SRTP according to RFC 3711 used to encrypt RTP and RTCP transport 1 3 2 MP 11x Hardware Features Combined FXS FXO gateways 4 FXS and 4 FXO ports on the MP 118 2 FXS and 2 FXO ports on the MP 114 MP 11x compact rugged enclosure only one half o
342. is either Relay or Mute The Valid range is 0 to 2 000 msec The default is 100 msec 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 The valid range is 0 to 2 000 msec The default is 1 000 msec SIP User s Manual 164 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 3 Restoring and Backing up the Gateway Configuration The Configuration File screen enables you to restore load a new ini file to the gateway or to back up make a copy of the VoIP gateway ini file and store it in a directory on your computer the current configuration the gateway is using Back up your configuration if you want to protect your VoIP gateway programming The backup ini file includes only those parameters that were modified and contain other than default values Restore your configuration if the VoIP gateway has been replaced or has lost its programming information you can restore the VoIP gateway configuration from a previous backup or from a newly created ini file To restore the VoIP gateway configuration from a previous backup you must have a backup of the VoIP gateway information stored on your computer gt To restore or back up the ini file take the following step E Open the Configuration File screen Advanced Configuration menu gt Configuration File the Configuration File screen is displayed Figure 5 45 Configurat
343. is 30 seconds The range is 10 to 3600 Determines the method the gateway uses to subscribe to an MWI server Per Endpoint 0 Each endpoint subscribes separately This method is usually used for FXS gateways default Per Gateway 1 Single subscription for the entire gateway This method is usually used for FXO gateways Disable 0 Disable default Using OPTIONS 1 Enable Keep alive with Proxy using OPTIONS Using REGISTER 2 Enable Keep alive with Proxy using REGISTER If EnableProxyKeepAlive 1 SIP OPTIONS message is sent every ProxyKeepAliveTime If EnableProxyKeepAlive 2 SIP REGISTER message is sent every RegistrationTime Any response from the Proxy either success 200 OK or failure 4xx response is considered as if the Proxy is correctly communicating Note 1 This parameter must be set to 1 OPTIONS when Proxy redundancy is used Note 2 When EnableProxyKeepAlive 2 REGISTER the homing redundancy mode is disabled Note 3 When the active proxy does not respond to INVITE messages sent by the gateway the proxy is marked 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 The default value is 60 seconds Note This parameter is applicable only if EnableProxyKeepAlive 1 OPTIONS When EnableProxyKeepAlive 2 REGISTER the time interval between Keep Alive messages is determined by
344. is displayed after pressing the button Submit if the entered value is incorrect Note 2 After changing the subnet mask and pressing the button Submit a prompt appears indicating that for the change to take effect the gateway is to be reset Default Gateway Address IP address of the default gateway used by the gateway Enter the IP address in dotted format notation for example 10 8 0 1 Note 1 A warning message is displayed after pressing the button Submit if the entered value is incorrect Note 2 After changing the default gateway IP address and pressing the button Submit a prompt appears indicating that for the change to take effect the gateway is to be reset For detailed information on multiple routers support refer to Section 9 6 on page 252 OAM Network Settings available only in Multiple IPs mode IP Address The gateway s source IP address in the OAM network LocalOAMIPAddress The default value is 0 0 0 0 Subnet Mask The gateway s subnet mask in the OAM network LocalOAMSubnetMask The default subnet mask is 0 0 0 0 Default Gateway Address N A LocalOAMDefaultGW Use the IP Routing table instead Advanced Configuration gt Network Settings Control Network Settings available only in Multiple IPs mode IP Address The gateway s source IP address in the Control network LocalControllPAddress The default value is 0 0 0 0 Subnet Mask The gateway s subnet mask in the Control network Local
345. is possible to demand the authentication of the client s certificate To enable two way authentication on the MediaPack set the ini file parameter SlPSRequireClientCertificate 1 For information on installing a client certificate refer to Section 12 2 5 on page 292 Embedded Web Server Configuration For additional security you can configure the Embedded Web Server to accept only secured HTTPS connections by changing the parameter HTTPSOnly to 1 described in Table 5 50 on page 174 You can also change the port number used for the secured Web server by default 443 by changing the ini file parameter HTTPSPort described in Table 5 55 on page 182 12 2 2 1 Using the Secured Embedded Web Server gt To use the secured Embedded Web Server take these 3 steps 1 Access the MediaPack using the following URL https host name or IP address Depending on the browser s configuration a security warning dialog may be displayed The reason for the warning is that the MediaPack initial certificate is not trusted by your PC The browser may allow you to install the certificate thus skipping the warning dialog the next time you connect to the MediaPack SIP User s Manual 288 Document LTRT 65408 SIP User s Manual 12 Security 2 If you are using Internet Explorer click View Certificate and then Install Certificate 3 The browser also warns you if the host name used in the URL is not identical to the one listed in the ce
346. it into the file or change it there accordingly Alias perl C Apache2 perl1 lt Location perl gt SetHandler perl script PerlResponseHandler ModPerl Registry Options ExecCGI PerlOptions ParseHeaders lt Location gt 8 Locate the file put cgi on the supplied software package and copy it into the Apach2 perl directory Change the first line in this file from c perl bin perl to your perl executable file this step is required only if mod_perl is not included in your Apache installation 9 Inthe apach2 bin directory from a DOS prompt type the following commands Apache exe n Apachez2 k install Apache exe n Apache2 k start The installation and configuration are finished You are now ready to start using the HTTP server G 2 Linux Operation Systems Note It is assumed that the installing of Linux already includes Apache server for example Apache 1 3 23 perl and mod_perl for example mod_perl 1 26 Additional required software an uploading script put cgi supplied with the software package gt To configure Apache HTTP server take these 4 steps 1 Inside the Apache directory create the directory perl for example var www perl Locate the file put cgi on the supplied software package and copy it to that directory 2 In the put cgi script change the first line from c perl bin perl to your perl executable file this step is required only if mod_perl is not included in your Apache install
347. its refer to Section C 11 on page 355 gt 1 To load the software and configuration files take these 4 steps Create a folder on your computer that contains all software and configuration files that are needed as part of the TFTP process Set the BootP and TFTP preferences refer to Section C 10 Add client configuration for the VoIP gateway that you want to initialize by the BootP refer to Section C 11 1 Reset the VoIP gateway either physically or remotely causing the device to use BootP to access the network and configuration information SIP User s Manual 350 Document LTRT 65408 SIP User s Manual C BootP TFTP Configuration Utility C 7 BootP TFTP Application User Interface Figure C 1 shows the main application screen for the BootP TFTP utility Figure C 1 Main Screen f 4 AudioCodes BootP TFTP Server File Services Edit Help Date Time Status New IP File Client Info 15 12 2003 10 22 30 100 OK D TFTPLoad MP108h323 ini 15 12 2003 1 22 100 OK D TFTPLoad ramMP108_H323 cmp 15 12 2003 1 22 100 OK D TFTPLoad TP161 0Sip ini 15 12 2003 1 22 100 OK D TFTPLoad TP161 0Sip ini 15 12 2003 1 22 Client Found 10 8 77 7 MP108_041 MP108 ip 10 8 77 7 boot 1 92 cmp 042002990361 FXS chnis 8 15 12 2003 21 4 100 0K D TFTPLoad ramTP1610_SIP cmp 15 12 2003 21 100 0K D TFTPLoadsramTP1610_SIP cmp Log Window C 8 Function Buttons on the Main Screen Pause Click this button to pause the
348. ivalence Number of all devices do not exceed five MP 11x FXO Notice The MP 11x FXO Output Tones and DTMF level should not exceed 9 dBm AudioCodes setting 23 in order to comply with FCC 68 TIA EIA IS 968 and TBR 21 The maximum allowed gain between any 2 ports connected to the PSTN should be set to 0 dB in order to comply with FCC 68 TIA EIA IS 968 Signal power limitation Network Compatibility of FXO Ports The products support the Telecom networks in EU that comply with TBR21 FCC Statement This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to part 15 of the FCC Rules These limits are designed to provide reasonable protection against harmful interference in a residential installation This equipment generates uses and can and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Connect the equipment into an outl
349. kControl must be set first followed by the acgwAdminState 14 11 2 Graceful Shutdown acgwAdminState is a read write MIB object When a get request is sent for this object the agent returns the board s current administrative state The possible values received on a get request include the following m locked 0 the board is locked E shuttingDown 1 the board is currently performing a graceful lock E unlocked 2 the board is unlocked On a set request the manager supplies one of the following desired administrative states E locked 0 E unlocked 2 When the board changes to either shuttingDown or locked state an adminStateChange alarm is raised When the board changes to an unlocked state the adminStateChange alarm is cleared Before setting acgwAdminState to perform a lock acgwAdminStateLockControl should be set first to control the type of lock that is performed The possible values for the acgwAdminStateLockControl include the following E 1 Perform a graceful lock Calls are allowed to complete No new calls are allowed to be originated on this device E 0 Perform a forced lock Calls are immediately terminated m Any number greater than 0 Time in seconds before the graceful lock turns into a forced lock Version 5 0 323 December 2006 Ta WH wt AudioCodes MediaPack 14 12 AudioCodes Element Management System Using AudioCodes Element Management System EMS is recommended to Custom
350. l side the Tel Source Number is used as the IP Source Number and also as the IP Display Name Overwrite 2 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 Version 5 0 61 December 2006 Ta WH wt AudioCodes MediaPack Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter Use Display Name as Source Number UseDisplayNameAsSourc eNumber Play Ringback Tone to IP PlayRBTone2IP Play Ringback Tone to Tel PlayRBTone2Tel Description Applicable to IP gt Tel calls No 0 The IP Source Number is used as the Tel Source Number and the IP Display Name is used as the Tel Display Name if IP Display Name is received If no Display Name is received from IP the Tel Display Name remains empty default Yes 1 If an IP Display Name is received it is used as the Tel Source Number and also as the Tel Display Name the 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 the Presentation is set to Restricted 1 For example When the following is received from 100 lt sip 200 201 202 203 204 gt the outgoing Source Number and Display Name are set to 100 and the Presentation is set to Allowed 0 When the following is received from lt sip 100 101 102 103 104 gt the outgoing Source Numbe
351. layed 4 Copy this text and send it to your security provider the security provider also known as Certification Authority or CA signs this request and send you a server certificate for the device 5 Save the certificate in a file e g cert txt Ensure the file is a plain text file with the BEGIN CERTIFICATE header The figure below is an example of a Base64 Encoded X 509 Certificate Figure 12 10 Example of a Base64 Encoded X 509 Certificate MIIDkzCCAnugAwI BAg IEAgAAADANBgkqhkiG9w0BAQOFADA MOswCOYDVOQQGEWJG UJETMBEGA1UEChHMKO2VydGlwb3N0ZTEbMBkGA1UEAxMSQ2VydGlwb3NO0ZSBTZXJ2 ZXVyMB4XDTk4MDY yNDA4MDAWMF oXDTE4MDY yNDA4MDAwMF owP zZELMAKGA1UEBhMC RLIxEzARBgNVBAoTCkN1icnRpcG9zdGUxGzAZBgNVBAMTEKN1cnRpcG9zdGUgU2Vy dmV 1c jCCASEwDOY JKoZ IhvcNAQEBBOADggEOADCCAQkCggEAPqd4MziR4spW1dGR x8bOrhZkonWnNm Yhb7 4067ecfl1janH7GcN SXsfx7jJprewUL 7v7Cvpr4R7qIi JcmdHIntmf 7JPM5n6cDBv17uSW63er7NkVnMFHWK10aGFLMybFkzaeGrvFm4k31R efixDmuOetFhJgHYezYHf44LvPRPwhSrzi9 Aq308pWDguJuZDIUP1F14jMa LPwv REX F CcUW w 6 Before continuing set the parameter HTTPSOnly 0 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 7 Inthe Certificates screen Figure 12 9 locate the server certificate loading section 8 Click Browse and navigate to the cert txt file click Send File 9 When the operation is completed save the configuration Section 5
352. lcos v 21 Modem Transport Type Disable A v 22 Modem Transport Type Enable Bypass A vz Modem Transport Type Enable Bypass A v 32 Modem Transport Type Enable Bypass A v 34 Modem Transport Type Enable Bypass A Fax Relay Redundancy Depth 2 Fax Relay Enhanced Redundancy Depth Fax Relay ECM Enable Enbe e Fax Relay Max Rate bps hao ooe Fax Modem Bypass Coder Type emaw oo Fax Modem Bypass Packing Factor e F CNG Detector Mode Disable 2 Configure the Fax Modem CID Settings according to Table 5 43 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 43 Media Settings Fax Modem CID Parameters continues on pages 154 to 157 Parameter Description Fax Transport Mode Fax Transport Mode that the gateway uses FaxTransportMode You can select Disable 0 transparent mode T 38 Relay 1 default Bypass 2 Events Only 3 Note If parameter IsFaxUsed 1 then FaxTransportMode is always set to 1 T 38 relay Caller ID Transport Type CallerIDTransportType NA SIP User s Manual 154 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 43 Media Settings Fax Modem CID Parameters continues on pages 154 to 157 Parameter Caller ID Type CallerIDType V 21 Modem Transport Type V21ModemTransportType V 22 Modem
353. le C 1 on page 359 Table 7 1 details the vendor specific information field according to device types Table 7 1 Vendor Specific Information Field Tag Description Value Length 220 Gateway Type 13 MP 124 1 14 MP 118 15 MP 114 16 MP 112 221 Current IP Address XXX XXX XXX XXX 4 222 Burned Boot Software Version X XX 4 223 Burned cmp Software Version XXXXXXXXXXXX 12 224 Geographical Address 0 31 1 225 Chassis Geographical Address 0 31 1 228 Indoor Outdoor 0 Indoor 1 Indoor is valid only for FXS 1 Outdoor FXO is always Outdoor 229 E amp M N A 1 230 Analog Channels 2 4 8124 1 Table 7 2 exemplifies the structure of the vendor specific information field for a TP 1610 slave module with IP address 10 2 70 1 Table 7 2 Structure of the Vendor Specific Information Field Vendor 4 4 4 lt lt lt lt 4 pe m m v m 2 ee D ov ov Specific 39 e 9 8 is e e Z Z Informati SQ Z Q amp 5 Zz alg ziken e e ee m on Code S 3 z 3 AS TOS O E 42 12 220 1 2 225 1 1 221 4 10 2 70 1 255 SIP User s Manual 214 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 8 1 8 1 1 8 1 1 1 Telephony Capabilities Working with Supplementary Services The MediaPack SIP FXS and FXO gateways support the following supplementary services Call Hold Retrieve refer to Section 8 1 1 on page 215 Consultation Alternate refer to Section 8 1 2 on page 216 Transf
354. le IPs network using the Embedded Web Server refer to Section 9 9 3 1 on page 257 and ini file refer to Section 9 9 3 2 on page 259 Table 9 2 below shows an example configuration that is implemented in the following sections Table 9 2 Example of VLAN and Multiple IPs Configuration Default ee IP Address Subnet Mask Gateway IP VLAN ID SAREE Address OAM 10 31 174 50 255 255 0 0 0 0 0 0 4 83 4 87 X Control 10 32 174 50 255 255 0 0 0 0 0 0 5 130 33 4 6 Media 10 33 174 50 255 255 0 0 10 33 0 1 6 Note that since a default gateway is available only for the Media network for the MediaPack to be able to communicate with an external device network on its OAM and Control networks IP routing rules must be used Note The values provided in Sections 9 9 3 1 and 9 9 3 2 are sample parameter values only and are to be replaced with actual values appropriate to your system SIP User s Manual 256 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities 9 9 3 1 Integrating Using the Embedded Web Server gt To integrate the MediaPack into a VLAN and multiple IPs network 4 using the Embedded Web Server take these 7 steps Access the Embedded Web Server Section 5 3 on page 51 Use the Software Upgrade Wizard Section 5 9 1 on page 197 to load and burn the firmware version to the MediaPack VLANs and multiple IPs support is available only when the firmware is burned to flash Configure the VLAN param
355. ler The parameter is used to maintain backward compatibility DTMF Transport Type DTMFTransportType MF Transport Type MFTransportType DTMF Volume 31 to 0 dB DTMFVolume Enable Answer Detector EnableAnswerDetector Answer Detector Activity Delay AnswerDetectorActivityDelay Answer Detector Silence Time AnswerDetectorSilenceTime Answer Detector Redirection AnswerDetectorRedirection Answer Detector Sensitivity AnswerDetectorSensitivity Description Voice gain control in dB This parameter sets the level for the transmitted IP gt Tel signal The valid range is 32 to 31 dB The default value is 0 dB PCM input gain control in dB This parameter sets the level for the received Tel gt IP signal The valid range is 32 to 31 dB The default value is 0 dB Note This parameter is intended for advanced users Changing it affects other gateway functionalities Disable 0 Silence Suppression disabled default Enable 1 Silence Suppression enabled Enable without adaptation 2 A single silence packet is sent during silence period applicable only to G 729 Silence Suppression is a method conserving bandwidth on VoIP calls by not sending packets when silence is detected Note If the selected coder is G 729 the following rules determine the value of the annexb parameter of the fmtp attribute in the SDP EnableSilenceCompression 0 gt annexb no EnableSilenceCompression
356. lient certificates cannot work gt To install a client certificate take these 6 steps 1 Before continuing set HTTPSOnly 0 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 2 Open the Certificates screen Advanced Configuration menu gt Security Settings submenu gt Certificates option the Certificates screen is displayed Figure 12 9 3 To load the Trusted Root Certificate file locate the trusted root certificate loading section 4 Click Browse and navigate to the file and then click Send File 5 When the operation is completed set the ini fie parameter HTTPSRequireClientCertificates 1 6 Save the configuration Section 5 10 2 on page 205 and restart the MediaPack When a user connects to the secure Web server m If the user has a client certificate from a CA listed in the Trusted Root Certificate file the connection is accepted and the user is prompted for the system password E 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 m fthe user doesn t have a client certificate from a listed CA or doesn t have a client certificate at all the connection is rejected The process of insta
357. line testing using the embedded Web server take these 2 steps 1 Click the Analog Line Testing submenu Status amp Diagnostics menu gt Analog Line Testing the Analog Line Testing confirmation box is displayed Figure 13 1 Analog Line Testing Confirmation Box Microsoft Internet Explorer j Line Testing will cause degradation to ongoing call on Channel 1 4re you sure you want to continue i Cancel 2 Click OK to confirm that you want to continue start the test the FXS Line Testing For Channel 1 screen appears Figure 13 2 FXS Line Testing For Channel 1 Screen Hook State On Hook Ring State Ring On Polarity Status Reverse On Message Waiting Indication Message Waiting Indication On Current Reading 10uA 85 Voltage Reading 10mv 5406 Analog Voltage Reading 10m 331 Ring Voltage Reading 10m 10856 Long Line Current Reading 10uA 195 To perform the test again click the ReTest button SIP User s Manual 302 Document LTRT 65408 SIP User s Manual 13 Diagnostics 13 3 Syslog Support Syslog protocol is an event notification protocol that enables a machine to send event notification messages across IP networks to event message collectors also known as Syslog servers Syslog protocol is defined in the IETF RFC 3164 standard Since each process application and operating system was written independently there is little uniformity to Syslog messa
358. ling network devices via SNMP E Trap Message The SNMP standard furnishes a mechanism by which devices can reach out to a Network Manager on their own via a trap message to notify or alert the manager of a problem with the device This typically requires each device on the network to be configured to issue SNMP traps to one or more network devices that are awaiting these traps The above message types are all encoded into messages referred to as Protocol Data Units PDUs that are interchanged between SNMP devices 14 1 2 SNMP MIB Objects The SNMP MIB is arranged in a tree structured fashion similar in many ways to a disk directory structure of files The top level SNMP branch begins with the ISO internet directory which contains the following four main branches mgmt SNMP branch Contains the standard SNMP objects usually supported at least in part by all network devices private SNMP branch Contains those extended SNMP objects defined by network equipment vendors experimental and directory SNMP branches Also defined within the internet root directory these branches are usually devoid of any meaningful data or objects The tree structure described above is an integral part of the SNMP standard though the most pertinent parts of the tree are the leaf objects of the tree that provide actual management data regarding the device Generally SNMP leaf objects can be partitioned i
359. lished call re INVITE Working with Redirect server and handling 3xx responses Early media supporting 183 Session Progress PRACK reliable provisional responses RFC 3262 Call Hold and Transfer Supplementary services using REFER Refer To Referred By Replaces and NOTIFY messages Supports RFC 3711 Secured RTP and Key Exchange according to lt draft ietf mmusic sdescriptions 12 gt Supports RFC 3489 Simple Traversal of UDP Through NATs STUN Supports RFC 3327 Adding Path to Supported header Supports RFC 3581 Symmetric Response Routing Supports RFC 3605 RTCP Attribute in SDP Supports RFC 3326 Reason header Supports RFC 4028 Session Timers in SIP Supports network asserted identity and privacy RFC 3325 and RFC 3323 Supports RFC 3911 The SIP Join Header Supports RFC 3903 SIP Extension for Event State Publication Supports RFC 3953 The Early Disposition Type for SIP Supports RFC 3966 The tel URI for Telephone Numbers Supports RFC 4244 An Extension to SIP for Request History Information Supports Tel URI Uniform Resource Identifier according to RFC 2806 bis Supports ITU V 152 Procedures for supporting Voice Band Data over IP Networks Remote party ID lt draft ietf sip privacy 04 txt gt Supports obtaining Proxy Domain Name s from DHCP Dynamic Host Control Protocol according to RFC 3361 Supports handling forking proxy multiple responses RFC 2833 Relay for DTMF Digits including paylo
360. livery feature enables sending of DTMF digits to the gateway s port after the line is offhooked FXS or seized FXO For IP gt Tel calls after the line is offhooked seized the MediaPack plays the DTMF digits of the called number towards the phone line Note 1 The called number can also include the characters p 1 5 seconds pause and d detection of dial tone If the character d is used it must be the first digit in the called number The character p can be used several times For example the called number can be as follows d1005 dpp699 p9p300 To add the d and p digits use the usual number manipulation rules Note 2 To use this feature with FXO gateways configure the gateway to work in one stage dialing mode Note 3 If the parameter EnableDigitDelivery is enabled it is possible to configure the gateway to wait for dial tone per destination phone number before or during dialing of destination phone number therefore the parameter IsWaitForDialTone that is configurable for the entire gateway is ignored Note 4 The FXS gateway sends 200 OK messages only after it finishes playing the DTMF digits to the phone line SIP User s Manual 78 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 8 Advanced Parameters General Parameters continues on pages 78 to 82 Parameter Enable DID Wink EnableDIDWink Reanswer Time RegretTime Delay
361. lling 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 SIP User s Manual 292 Document LTRT 65408 SIP User s Manual 12 Security 12 3 SRTP The gateway supports Secured RTP SRTP according to RFC 3711 SRTP is used to encrypt RTP and RTCP transport since it is best suited for protecting VoIP traffic SRTP requires a Key Exchange mechanism that is performed according to lt draft ietf mmusic sdescriptions 12 gt The Key Exchange is executed by adding a Crypto attribute to the SDP This attribute is used by both sides to declare the various supported cipher suites and to attach the encryption key to use If negotiation of the encryption data is successful the call is established Use the parameter MediaSecurityBehaviour described in Table 5 50 to select the gateway s mode of operation Must or Prefer These modes determine the behavior of the gateway if negotiation of the cipher suite fails m Mandatory the call is terminated Incoming calls that don t include encryption information are rejected m Preferable an unencrypted call is established Incoming calls that don t include encryption information are accepted To enable SRTP set the parameter EnableMed
362. lnfoFileName described in Table 5 64 the Embedded Web Server refer to Section 5 9 2 on page 202 or by using the automatic update mechanism UserlnfoFileURL refer to Section 10 3 on page 263 Table 15 1 User Information Items Item Description Maximum Size PBX extension The relevant PBX extension number 10 Global IP The relevant IP phone number 20 A string that represents the PBX extensions pepley name for the Caller ID a0 Username A string that represents the username for 20 SIP registration Password A string that represents the password for 20 SIP registration Figure 15 5 Example of a User Information File UserInformationFile1000 txt Notepad 4 File Edit Format Help 401 6380001 DN401 UN401 401 408 6380008 DN408 UN408 401 SIP User s Manual 332 Document LTRT 65408 SIP User s Manual 16 Selected Technical Specifications 16 Selected Technical Specifications 16 1 MP 11x Specifications Table 16 1 MP 11x Functional Specifications continues on pages 333 to 335 Channel Capacity Available Ports MP 112 2 ports MP 114 4 ports MP 118 8 ports The MP 112 differs from the MP 114 and MP 118 Its configuration excludes the RS 232 connector the Lifeline option and outdoor protection MP 11x FXS Functionality FXS Capabilities Short or Long Haul Automatic Detection REN2 Up to 10 km 32 800 feet using 24 AWG line RENS Up to 3 5 km 11 400 feet using 24 AWG line Note
363. load screen is displayed shown in Figure 10 7 Click the Browse button in the Send Background Image File from your computer to gateway box Navigate to the folder that contains the background image file you want to load Click the Send File button the file is sent to the device When loading is complete the screen is automatically refreshed and the new background image is displayed To save the image to flash memory so it is available after a power fail refer to Section 5 10 2 on page 205 The new background appears on all Web Interface screens 2 If you encounter any problem during the loading of the files or you want to restore the default images click the Restore Default Images button When replacing both the background image and the logo image first load the logo image followed by the background image To replace the background image via the ini file take these 2 steps Place your background image file in the same folder as where the device s ini file is located i e the same location defined in the BootP TFTP configuration utility For detailed information on the BootP TFTP refer to Appendix C on page 349 Add modify the ini file parameters in Table 10 5 according to the procedure described in Section 6 2 on page 209 Note that loading the device s ini file via the Configuration File screen in the Web Interface doesn t load the logo image file as well Table 10 5 Customizable Logo ini File Para
364. lowing dimensions e Side to side distance 140 mm e Front to back distance 101 4 mm Insert a wall anchor of the appropriate size into each hole Fasten a DIN 96 3 5X20 wood screw not supplied into each of the wall anchors Position the four oval notches located on the base of the MP 11x refer to item 2 in Figure 3 2 over the four screws and hang the MP 11x on them Version 5 0 31 December 2006 Ta Ao wi AudioCodes MediaPack 3 1 4 3 Installing the MP 11x in a 19 inch Rack The MP 11x can be installed in a standard 19 inch rack by placing it on an AudioCodes 19 inch rack mounting shelf that is pre installed in the rack The shelf can hold up to two MP 11x gateways This shelf can be ordered separately from AudioCodes Note The 19 inch rack shelf is not supplied in the standard package kit but can be ordered separately Bulk Pack package MCMKO00015 with 10 rack mounting shelves for MP 11x For ordering and pricing please contact your AudioCodes distributor Figure 3 3 MP 11x Rack Mount Table 3 2 MP 11x Rack Mount Item Functionality 1 Standard rack holes used to attach the shelf to the rack 2 Eight shelf to device screws gt To install the MP 11x in a 19 inch rack take these 3 steps 1 Use the shelf to device screws supplied to attach one or two MP 11x devices to the shelf 2 Position the shelf in the rack and line up its side holes with the rack frame holes 3 Use four standard ra
365. m a preconfigured TFTP server If a preconfigured TFTP server doesn t exist the gateway operates using the existing software and configuration files loaded on its non volatile memory Note that after the operational software runs if DHCP is configured the gateway attempts to renew its lease with the DHCP server Though DHCP and BootP servers are very similar in operation the DHCP server includes some differences that could prevent its operation with BootP clients However many DHCP servers such as Windows NT DHCP server are backward compatible with BootP protocol and can be used for gateway configuration The time duration between BootP DHCP requests is set to 1 second by default This can be changed by the BootPDelay ini file parameter Also the number of requests is 3 by default and can be changed by BootPRetries ini file parameter both parameters can also be set using the BootP command line switches Version 5 0 265 December 2006 Ta Ce AudioCodes Figure 10 3 MediaPack Startup Process Physical Reset Reset from the Web Interface or SNMP BootP N x times a o Response Response Update network BootP Response DHCP Response p parameters from p BootP DHCP reply BootP DHCP reply contains firmware file name Download firmware via TFTP BootP DHCP reply contains ini file name BootP DHCP reply contains ini file name Preconfigured Download configuration files via TFTP Ru
366. maintains a history of alarms that have been raised and traps that have been cleared to allow a manager to recover any lost raised or cleared traps Two views of the alarm history table are supported by the agent E acAlarmHistoryTable in the enterprise acAlarm E nilmLogTable and nlmLogVariableTable in the standard NOTIFICATION LOG MIB As with the acActiveAlarmTable the acAlarmHistoryTable is a simple one row per alarm table that is easy to view with a MIB browser Version 5 0 307 December 2006 7a Wa wt AudioCodes MediaPack 14 3 14 4 Cold Start Trap MediaPack technology supports a cold start trap to indicate that the device is starting This allows the manager to synchronize its view of the device s active alarms Two different traps are sent at start up m The standard coldStart trap iso 1 org 3 dod 6 internet 1 snmpV2 6 snmpModules 3 snmpMIB 1 snmpMIBObjects 1 snmpTraps 5 coldStart 1 sent at system initialization m The enterprise acBoardEvBoardStarted which is generated at the end of system initialization This is more of an application level cold start sent after the entire initializing process is complete and all the modules are ready Third Party Performance Monitoring Measurements Performance measurements are available for a third party performance monitoring system through an SNMP interface These measurements can be polled at scheduled intervals by an external poller or utility in
367. mber of UDP transmissions first transmission retransmissions of SIP messages The range is 1 to 7 The default value is 7 Enables disables the usage of the SIP Reason header 0 Disable 1 Enable default SIP User s Manual 64 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 1 2 Proxy amp Registration Parameters Use this screen to configure parameters that are associated with Proxy and Registration gt To configure the Proxy amp Registration parameters take these 4 steps 1 Open the Proxy amp Registration parameters screen Protocol Management menu gt Protocol Definition submenu gt Proxy amp Registration option the Proxy amp Registration parameters screen is displayed Figure 5 6 Proxy amp Registration Parameters Screen Proxy amp Registration Enable Proxy Use Proxy Proxy Name Proxy IP Address 10 2 1 2 First Redundant Proxy IP Address 0 0 0 0 Second Redundant Proxy IP Address 0 0 0 0 Third Redundant Proxy IP Address 0 0 0 0 Redundancy Mode Parking Proxy Load Balancing Method Disable Proxy IP List Refresh Time 60 Enable Proxy Keep Alive Disable Proxy Keep Alive Time 60 Enable Fallback to Routing Table Disable Prefer Routing Table No Use Routing Table for Host Names and Profiles Disa
368. me_1 g 711Ulaw64k 40 CoderName_1 g 231 90 1 1 CoderName_2 g726 2 0 SIP User s Manual 110 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 6 2 Tel Profile Settings Use the Tel Profile Settings screen to define up to four different Tel Profiles These Profiles are used in the Endpoint Phone Number table to associate different Profiles to gateway s endpoints thereby applying different behavior to different MediaPack ports gt To configure the Tel Profile settings take these 9 steps 1 Open the Tel Profile Settings screen Protocol Management menu gt Profile Definitions submenu gt Tel Profile Settings option the Tel Profile Settings screen is displayed Figure 5 23 Tel Profile Settings Screen Tel Profile Settings Profile ID 1 v Profile Name Default Tel Profile Profile Parameters Profile Preference 1 v Fax Signaling Method T 38 Relay Dynamic Jitter Buffer Minimum Delay msec Dynamic Jitter Buffer Optimization Factor RTP IP Diff Serv Signaling DiffServ Voice Volume 32 to 31 dB DTMF Volume 31 to 0 dB Input Gain 32 to 31 dB Enable Polarity Reversal Disable Enable Current Disconnect Disable Enable Digit Delivery Disable MW Analog Lamp Disable MWI Display Disable Echo Canceler Enable Max Hook Flash Detection Perio
369. ment 3 4 5 6 Dest Phone Prefix Source Phone Prefix Dest IP Address Profile ID Status Charge Code Tel to IP routing can be performed either before or after applying the number manipulation rules To control when number manipulation is done set the Tel to IP Routing Mode parameter described in Table 5 17 To configure the Tel to IP Routing table take these 6 steps Open the Tel to IP Routing screen Protocol Management menu gt Routing Tables submenu gt Tel to IP Routing option the Tel to IP Routing screen is displayed shown in Figure 5 17 In the Tel to IP Routing Mode field select the Tel to IP routing mode refer to Table 5 17 In the Routing Index drop down list select the range of entries that you want to edit Configure the Tel to IP Routing table according to Table 5 17 Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Figure 5 17 Tel to IP Routing Table Screen 10 fi 00 10 33 45 63 1 OK 10 33 45 60 1 QOS Low 10 33 45 64 1 ok i domain com 1 Dns Error 0 0 0 0 2 pe a 10 13 77 7 1 Ping Error 13 77 Oo 0j ns oO oy ee wj Nj oOo Table 5 17 Tel to IP Routing Table continues on pages 101 to 102 Parameter Description Tel to IP Routing Mode Route calls before manipulation 0 Tel gt IP calls are routed before the number
370. meters Description BkglmageFileName The name and path of the file containing the new background Use a gif jpg or jpeg image file The default is AudioCodes background file Note The length of the name of the image file is limited to 47 characters SIP User s Manual 274 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 6 3 Customizing the Product Name The Product Name text string can be modified according to OEMs specific requirements mH To replace AudioCodes default product name with a text string via the Web Interface modify the two ini file parameters in Table 10 6 according to the procedure described in Section 10 6 4 on page 276 m To replace AudioCodes default product name with a text string via the ini file add modify the two ini file parameters in Table 10 6 according to the procedure described in Section 6 2 on page 209 Table 10 6 Web Appearance Customizable ini File Parameters Parameter Description UseProductName 0 Don t change the product name default 1 Enable product name change UserProductName Text string that replaces the product name The default is MediaPack The string can be up to 29 characters Version 5 0 275 December 2006 7a Ta P wt AudioCodes MediaPack 10 6 4 Modifying ini File Parameters via the Web AdminPage gt To modify ini file parameters via the AdminPage take these 6 steps Access the MediaPack Embedded Web Server
371. meters via the ini file Channel parameters are changeable on the fly Changes take effect from next call 5 6 2 1 Configuring the Voice Settings gt To configure the Voice Settings parameters take these 4 steps 1 Open the Voice Settings screen Advanced Configuration menu gt Media Settings gt Voice Settings option the Voice Settings screen is displayed Figure 5 40 Voice Settings Screen Voice Settings Voice Volume 32 to 31 dB Input Gain 32 to 31 dB Silence Suppression Disable Echo Canceler On z DTMF Transport Type RFC2833 Relay DTMF MF Transport Type RFC2833 Relay MF DTMF Volume 31 to 0 dB 11 Enable Answer Detector Disable Answer Detector Activity Delay Answer Detector Silence Time Answer Detector Redirection Disable Answer Detector Sensitivity fo SIP User s Manual 152 Document LTRT 65408 SIP User s Manual 5 Web Management 2 Configure the Voice Settings according to Table 5 42 9 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 42 Media Settings Voice Settings Parameters Parameter Voice Volume VoiceVolume Input Gain InputGain Silence Suppression EnableSilenceCompression The parameter is used to maintain backward compatibility Echo Canceler EnableEchoCancel
372. method to aggregate RTP streams from several channels to reduce the bandwidth overhead caused by the attached Ethernet IP UDP and RTP headers and to reduce 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 ThroughPacket is accomplished by aggregating payloads from several channels that are sent to the same destination IP address into a single IP packet ThroughPacket can be applied to the entire gateway or using IP Profile to specific IP destinations refer to Section 5 5 6 3 on page 113 Note that ThroughPacket must be enabled on both gateways To enable ThroughPacket set the parameter RemoteBaseUDPPort to a nonzero value Note that the value of RemoteBaseUDPPort on the local gateway must equal the value of BaseUDPPort of the remote gateway The gateway uses these parameters to identify and distribute the payloads from the received multiplexed IP packet to the relevant channels In ThroughPacket mode the gateway 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 ThroughPacket is used Call statistics aren t available since there is no RTCP flow Version 5 0 229 December 2006 Ta WH wt AudioCodes MediaPack 8 6 Dynamic Jitter Buff
373. method used between the PBX and the gateway Valid options include None default 0 DTMF 1 SMDI 2 SIP User s Manual 130 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 34 Voice Mail Parameters continues on pages 130 to 131 Parameter Description Line Transfer Mode LineTransferMode Digit Patterns Determines the transfer method used by the gateway Disable 0 IP default Blind Transfer 1 PBX blind transfer In this mode 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 The following digit pattern parameters apply only to VM applications that use the DTMF communication method For the available patterns syntaxes refer to the CPE Configuration Guide for Voice Mail Forward on Busy Digit Pattern DigitPatternForwardOnBus y Forward on No Answer Digit Pattern DigitPatternForwardOnNoA nswer Forward on Do Not Disturb Digit Pattern DigitPatternForwardOnDN D Forward on No Reason Digit Pattern DigitPatternForwardNoRea son Internal Call Digit Pattern DigitPatterninternalCall External Call Digit Pattern DigitPatternExternalCall Disconnect Call Digit Pattern TelDisconnectCode MWI MWI Off Digit Pattern MWIOffCode MWI On Digit Pattern MWIOnCode SMDI
374. n Defines the mode of operation when the 3 Way Conference feature is used Valid options include 0 Conference initiating INVITE sent by the gateway uses the ConferencelD concatenated with a unique identifier as the Request UR default 1 Conference initiating INVITE sent by the gateway uses only the ConferencelD as the Reques URI If 3wayConferenceMode is set to 0 the Conference initiating INVITE sent by the gateway 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 If 3wayConferenceMode is set to 1 the Conference initiating INVITE sent by the gateway only uses the ConferencelD as the Reques URI The media 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 gateway in the Refer To header value in the REFER messages sent by the gateway to the remote parties The remote parties join the conference by sending INVITE messages to the media server using this conference URI Defines the duration in seconds for which Off Hook Warning Tone is played to the user The valid range is 1 to 2 147 483 647 seconds The default is 600 seconds Note A negative value indicates that the tone is played infinitely Defines the number of rings before the FXO gateway answers a
375. n For example the entry 20 29 means all numbers in the range 20 to 29 Searching for Configuration Parameters The Embedded Web Server provides a search engine that allows you to search any ini file parameter that is configurable by the Web server The Search button located near the bottom of the Main menu bar refer to Figure 5 2 on page 52 is used to perform parameter searches 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 the Embedded Web Server lists all parameters that contain the searched sub string in their parameter names gt To search for an ini file parameter configurable by the Web server take these 3 steps 1 In the Search Engine field enter the ini parameter name or sub string of the parameter name 2 Click Search The Searched Result screen appears listing all searched parameter results Figure 5 3 Searched Result Screen fd AudioCodes MP 118 FXS A Quick Setup Found 17 pararr searching for sec 0 602000 seconds Protocol Management DEFAULTRELEASECAUSE Advanced Configuration The release cause that will be sent to IP or to Tel when gateway initiates release by itself Status amp Diagnostics Link GeneralParameters Software Update Maintenance DNSSecServerIP Log Off Defines the DNS Secondary server IP address Link PSettings ENABLEIPSEC This parameter is used
376. n sec MaxCallDuration Enable LAN Watchdog EnableLanWatchDog Description No 0 Busy out feature is not used default Yes 1 Busy out feature is enabled When Busy out is enabled the MediaPack gateway performs a specific behavior e g plays a reorder tone when the phone is offhooked due to one of the following reasons Physically disconnected from the network i e Ethernet cable is disconnected The Ethernet cable is connected but the gateway can t communicate with any host Note that LAN Watch Dog must be activated EnableLANWatchDog 1 The gateway can t communicate with the gatekeeper proxy and no other alternative exists to send the call Note The FXSOOSBehavior parameter is used to control the behavior of the FXS endpoints of the gateway when a Busy out or Graceful Lock occurs Note FXO endpoints during Busy out and Lock are inactive Note Refer to LifeLineType parameter for complementary optional behavior Default Release Cause to IP for IP gt Tel calls used when the gateway initiates a call release and if an explicit matching cause for this release isn t found a default release cause can be configured The default release cause is NO_ ROUTE_TO_DESTINATION 3 Other common values are NO_CIRCUIT_AVAILABLE 34 DESTINATION_OUT_OF_ORDER 27 etc Note The default release cause is described in the Q 931 notation and is translated to corresponding SIP 40x or 50x value
377. n before the call is routed to the corresponding hunt group the prefix 21 is removed from the original number so that only 100 is left Note 1 Applicable only if number manipulation is performed after call routing for IP gt Tel calls refer to IP to Tel Routing Mode parameter Note 2 Similar operation of removing the prefix is also achieved by using the usual number manipulation rules No 0 Disable the Alternative Routing feature default Yes 1 Enable the Alternative Routing feature Status Only 2 The Alternative Routing feature is disabled A read only information on the quality of service of the destination IP addresses is provided For information on the Alternative Routing feature refer to Section 8 7 on page 231 None 0 Alternative routing is not used Conn 1 Alternative routing is performed if ping to initial destination failed Qos 2 Alternative routing is performed if poor quality of service was detected Both 3 Alternative routing is performed if either ping to initial destination failed or poor quality of service was detected or DNS host name is not resolved default Note QoS Quality of Service is quantified according to delay and packet loss calculated according to previous calls QoS statistics are reset if no new data is received for two minutes For information on the Alternative Routing feature refer to Section 8 7 on page 231 Packet loss percentage at which the IP
378. n each row in the srCommunityTable to the empty string 2 Remove the appropriate row from the snmpTargetAddrTable The change takes effect immediately All managers can now access the device SNMP Ports The SNMP Request Port is 161 and the Trap Port is 162 These ports can be changed by setting parameters in the device ini file The parameter name is SNMPPort lt port_number gt Valid UDP port number default 161 This parameter specifies the port number for SNMP requests and responses Usually it should not be specified Use the default Multiple SNMP Trap Destinations An agent can send traps to up to five managers For each manager set the manager s IP address receiving port number and enable sending traps to that manager The user also has the option of associating a trap destination with a specific SNMPv3 USM user Traps are then sent to that trap destination using the SNMPv3 format and the authentication and privacy protocol configured for that user To configure the trap managers table use m The Embedded Web Server refer to Section 5 6 6 1 on page 178 E The inifile refer to Section 14 8 1 1 below E SNMP refer to Section 14 8 1 3 on page 314 Configuring Trap Manager via Host Name One of the five available SNMP managers can be defined using a FQDN In the current version this option can only be configured via the ini file SNMPTrapManagerHostName The gateway tries to resolve the host name at start up O
379. n internal routing table Fallback to internal routing table if Proxy is not responding Supports up to four Proxy servers If the primary Proxy fails the gateway automatically switches to a redundant Proxy Supports domain name resolving using DNS NAPTR and SRV records for Proxy Registrar and domain names that appear in the Contact and Record Route headers Supports Load Balancing over Proxy servers using Round Robin or Random Weights Proxy or Registrar Registration such as REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z9hG4bRaC7AU2 34 From lt Sip GWRegistrationName sipgatewayname gt tag 1c29347 To lt sip GWRegistrationName sipgatewayname gt Call ID 10453 212 179 22 229 Seq 1 REGISTER Expires 3600 Contact sip GWRegistrationName 212 179 22 229 Content Length 0 e The servername string is defined according to the following rules e The servername is equal to RegistrarName if configured The RegistrarName can be any string e Otherwise the servername is equal to RegistrarlP either FQDN or numerical IP address if configured e Otherwise the servername is equal to ProxyName if configured The ProxyName can be any string e Otherwise the servername is equal to ProxylP either FQDN or numerical IP address The parameter GWRegistrationName can be any string If the parameter is not defined the parameter UserName is used instead The sipgatewayname parameter
380. n is available for installing the MP 11x in a 19 inch rack The 19 inch rack installation package contains a single shelf shown in Figure 3 1 and eight shelf to device screws Note The 19 inch rack shelf is not supplied in the standard package kit but can be ordered separately Bulk Pack package MCMK00015 containing 10 rack mounting shelves for MP 11x For ordering and pricing please contact your AudioCodes distributor Figure 3 1 19 inch Rack Shelf 3 1 4 Mounting the MP 11x The MP 11x can be mounted on a desktop refer to Section 3 1 4 1 below on a wall refer to Section 3 1 4 2 or installed in a standard 19 inch rack refer to Section 3 1 4 3 Figure 3 2 below describes the MP 11x base Figure 3 2 View of the MP 11x Base SIP User s Manual 30 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack Item Table 3 1 View of the MP 11x Base Functionality Square slot used to attach anti slide bumpers for desktop mounting Oval notch used to attach the MP 11x to a wall Screw opening used to attach the MP 11x to a 19 inch shelf rack 3 1 4 1 Mounting the MP 11x on a Desktop Attach the four supplied anti slide bumpers to the base of the MP 11x refer to item 1 in Figure 3 2 and place it on the desktop in the position you require 3 1 4 2 Mounting the MP 11x on a Wall gt 1 w To mount the MP 11x on a wall take these 4 steps Drill four holes according to the fol
381. n is issued The valid range is 1 to 30 The default value is 10 Determine 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 Requires client certificates for HTTPS connection The client certificate must be preloaded to the gateway and its matching private key must be installed on the managing PC Time and date must be correctly set on the gateway for the client certificate to be verified 0 Client certificates are not required default 1 Client certificates are required Defines the name of the HTTPS trusted root certificate file to be loaded via 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 relevant when the gateway is loaded via BootP TFTP For information on loading this file via the Embedded Web Server refer to the Security section in the User s 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 via TFTP The file must be in base64 encoded PEM format The valid range is a 47 character string Note This parameter is only relevant when the gateway is loaded via BootP TFTP For information on loading this file via the Embedded Web Server refer to the Security section in the User s Manual
382. n it with an additional entry in the IP to Hunt Group Routing table repeat the same routing rules with a different hunt group ID For detailed information on the Reasons for Alternative Routing Tables refer to Section 5 5 5 5 on page 106 SIP User s Manual 102 Document LTRT 65408 SIP User s Manual 5 Web Management To use hunt groups you must also do the following You must assign a hunt group ID to the VoIP gateway channels on the Endpoint Phone Number screen For information on how to assign a hunt group ID to a channel refer to Section 5 5 7 on page 115 You can configure the Hunt Group Settings table to determine the method in which new calls are assigned to channels within the hunt groups a different method for each hunt group can be configured For information on how to enable this option refer to Section 5 5 8 on page 117 If a Channel Select Mode for a specific hunt group isn t specified then the global Channel Select Mode parameter defined in General Parameters screen under Advanced Parameters applies To configure the IP to Hunt Group Routing table take these 6 steps Open the IP to Hunt Group Routing screen Protocol Management menu gt Routing Tables submenu gt IP to Hunt Group Routing option the IP to Hunt Group Routing table screen is displayed shown in Figure 5 18 Figure 5 18 IP to Hunt Group Routing Table Screen Dest Phone Prefix Source Phone Prefi
383. n number prefix h Notin use should be set to l Source IP address obtained from the Request URI in the INVITE message The b to d manipulation rules are applied if the called and calling numbers match the a g and I conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example NumberMapIP2Tel 01 2 972 034 NumberMapIP2Tel 03 2 667 22 Note The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The wildcard represents any number between 0 and 255 e g 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 Dialing Plan Notation The dialing plan notation applies in addition to the four Manipulation tables also to Tel gt IP Routing table and to P gt Hunt Group Routing table When entering a number in the destination and source Prefix columns you can create an entry that represents multiple numbers using the following notation E n m represents a range of numbers E n m represents multiple numbers Note that this notation only supports single digit numbers E x represents any single digit E that terminates the number represents the end of a number E A single asterisk represents any number For example 5551200 5551300
384. n operational software firmware URL Download firmware via TFTP Device Reset Preconfigured ini file URL SIP User s Manual 266 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 5 Using Parameter Tables The MediaPack uses parameter tables to group related parameters of specific entities and manage them together These tables similar to regular parameters can be configured via the ini file Embedded Web Server SNMP etc Tables are composed of lines and columns Columns represent parameters types Lines represent specific entities The instances in each line are called line attributes Lines in table may represent for example a trunk an NFS file system list of timers for a given application etc Table 10 1 and Table 10 2 below provide useful examples for reference Table 10 1 Example of Parameter Table Remote Management Connections Index Fields 1 Connection Number Connection Time Connected nee Number User Name User Password msec Permissions 0 Admin Yellow9 0 All 1 Gillian Red5 1266656 Read Only 2 David Orange6 0 Read Write Table 10 2 Example of Parameter Table Port to Port Connections Index Fields 1 Source Ports 2 Destination IP 3 Destination Port Source Port Destination IP Destination Port Connection Name Application Type 2020 10 4 1 50 2020 ATM_TEST_EQ LAB_EQ 2314 212 199 201 20 4050 ATM_ITROP_LOOP LAB_EQ 6010 10 3 3 41 6010 REMOTE_MGMT MGMT
385. n the Priority Weight and Port fields enter the relevant values 6 Repeat steps 4 to 5 for the second and third DNS names if required 7 Repeat steps 2 to 6 for each Internal SRV Table entry 8 Click the Submit button to save your changes 9 To save the changes so they are available after a hardware reset or power fail refer to Section 5 10 2 on page 205 Table 5 20 Internal SRV ini File Parameter Parameter Name in ini File Parameter Format SRV2IP SRV2IP lt Internal Domain Name gt lt Transport Type gt lt DNS Name 1 gt lt Priority 1 gt lt Weight 1 gt lt Port 1 gt lt DNS Name 2 gt lt Priority 2 gt lt Weight 2 gt lt Port 2 gt lt DNS Name 3 gt lt Priority 3 gt lt Weight 3 gt lt Port 3 gt Note 1 If the internal SRV table is configured the gateway first tries to resolve a domain name using this table If the domain name isn t found the gateway performs an SRV resolution using an external DNS server Note 2 This parameter can appear up to 10 times SIP User s Manual 106 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 5 6 Reasons for Alternative Routing The Reasons for Alternative Routing screen includes two tables TelSIP and IP gt Tel Each table enables you to define up to 4 different release reasons If a call is released as a result of one of these reasons the gateway tries to find an alternative route to that call The release
386. n the signal level sent to the hybrid and the echo level returning from the hybrid 0 6 dB default 1 9 dB 2 0 dB 3 3 dB FaxModemRelayVolume 18 to 3 corresponding to 18 dBm to 3 dBm in 1 dB steps Default 12 dBm fax gain control MGCPDTMFDetectionPoint 0 DTMF event is reported on the end of a detected DTMF digit 1 DTMF event is reported on the start of a detected DTMF digit default DTMFDigitLength Time in msec for generating DTMF tones to the PSTN side if TxDTMF Option 1 2 or 3 The default value is 100 msec The valid range is 0 to 32767 DTMFinterDigitinterval Time in msec between generated DTMF digits to PSTN side if TxDTMF Option 1 2 or 3 The default value is 100 msec The valid range is 0 to 32767 TestMode 0 CoderLoopback encoder decoder loopback inside DSP 1 PCMLoopback loopback the incoming PCM to the outgoing PCM 2 Tonelnjection generates a 1000 Hz tone to outgoing PCM 3 NoLoopback default ModemBypassPayloadType Modem Bypass dynamic payload type The valid range is 0 to 127 The default value is 103 BellModemTransportType Determines the Bell modem transport method 0 Transparent default 2 Bypass 3 Transparent with events FaxModemBypassBasicRtpPa Determines the basic frame size that is used during fax modem bypass cketinterval sessions 0 set internally default 1 5 msec 2 10 msec 3 20 msec Note When set for 5 m
387. nce of service may be required If advance notice is not practical you will be notified as soon as possible Also you will be advised of your right to file a compliant with the FCC if it is necessary 5 The telephone company may make changes in its facilities equipment operations or procedures that could affect the operation of the equipment If this happens the telephone company will provide advance notice in order for you to make necessary modifications to maintain uninterrupted service 6 If trouble is experienced with this equipment for repair or warranty information please contact AudioCodes Inc 2099 Gateway Place Suite 500 San Jose CA 95110 phone number 1 408 441 1175 If the equipment is causing harm to the telephone network the telephone company may request to disconnect the equipment until the problem is resolved 7 Installation is described in the Product User s manual Connection to Telephone Company provided coin service is prohibited Connection to party lines service is subject to State tariffs Version 5 0 389 December 2006 gt vee w AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 5 0 fa AudioCodes www audiocodes com THE STANDARDS INSTITUTION OF ISRAEL
388. nce the name is resolved IP is found the resolved IP address replaces the last entry in the trap manager table defined by the parameter SNMPManagerTablelP_x and the last trap manager entry of snmpTargetAddrTable in the snmpTargetMIB The port is 162 unless specified otherwise the row is marked as used and the sending is enabled When using host name resolution any changes made by the user to this row in either MIBs are overwritten by the gateway when a resolving is redone once an hour Note that several traps may be lost until the resolving is complete Version 5 0 319 December 2006 Ta ful AudioCodes MediaPack 14 8 5 2 Configuring Trap Managers via the ini File In the MediaPack ini file the parameters below can be set to enable or disable the sending of SNMP traps Multiple trap destinations can be supported on the device by setting multiple trap destinations in the ini file SNMPManagerTrapSendingEnable_ lt x gt indicates whether or not traps are to be sent to the specified SNMP trap manager A value of 1 means that it is enabled while a value of 0 means disabled lt x gt represents a number 0 1 2 which is the array element index Currently up to five SNMP trap managers can be supported SNMPManagerTrapUser_ lt x gt indicates to send an SNMPvz2 trap using the trap user community string configured with the SNMPTrapCommunityString parameter The user may instead specify an SNM
389. nclude Disable 0 default Enable 1 The gateway obtains a GRUU by generating a normal REGISTER request This request contains a Supported header field with the value gruu The gateway includes a tsip instance Contact header field parameter for each contact for which the GRUU is desired This Contact parameter contains a globally unique ID that identifies the gateway instance The global unique id is as follows If registration is per endpoint AuthenticationMode 0 it is the MAC address of the gateway concatenated with the phone number of the endpoint Ifthe registration is per gateway AuthenticationMode 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 gateway 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 opaque URI parameter the gateway obtains
390. nd can enable disable ECM fax mode using the FaxRelayECMEnable parameter When using T 38 mode the user can define a redundancy feature to improve fax transmission over congested P network This feature is activated by FaxRelayRedundancyDepth and FaxRelayEnhancedRedundancyDepth parameters Although this is a proprietary redundancy scheme it should not create problems when working with other T 38 decoders Note T 38 mode currently supports only the T 38 UDP syntax 8 3 3 Configuring Fax Modem Bypass Mode When VxxTransportType 2 FaxModemBypass Vxx can be one of the following V32 V22 Bell V34 Fax then on detection of fax modem the channel automatically switches from the current voice coder to a high bit rate coder G 711 or G 726 as defined by the user with the FaxModemBypassCoderType configuration parameter During the bypass period the coder uses the packing factor by which a number of basic coder frames are combined together in the outgoing WAN packet set by the user in the FaxModemBypassM configuration parameter 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 When fax modem transmission ends the reverse switching from bypass coder to regular voice coder is carried out SIP User s Manual 220 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 3 4 8 3 4 1
391. nd 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 ThroughPacket is disabled Note To enable ThroughPacket the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort must be set to a non zero value Determines the local UDP port used for outgoing multiplexed RTP packets applies to the ThroughPacket mechanism The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 ThroughPacket is disabled This parameter cannot be changed on the fly and requires a gateway reset Determines the remote UDP port the multiplexed RTP packets are sent to and the local UDP port used for incoming multiplexed RTP packets applies to the ThroughPacket mechanism The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 ThroughPacket is disabled This parameter cannot be changed on the fly and requires a gateway reset Note All gateways that participate in the same ThroughPacket session must use the same L1L1ComplexRxUDPPort Enables negotiation and usage of Comfort Noise CN Valid options include 0 Disable default 1 Enable Comfort Noise negotiation The use of CN is indicated by including a payload type for CN on the media description line of the SDP The gateway can use CN with a codec whose
392. ndard DES 3DES and Advanced Encryption Standard AES Hash types for IKE SA SHA1 and MD5 IPSec IPSec is responsible for encrypting and decrypting the IP streams The IPSec Security Policy Database SPD table defines up to 20 IP peers to which the IPSec security is applied IPSec can be applied to all packets designated to a specific IP address or to a specific IP address port source or destination and protocol type Each outgoing packet is analyzed and compared to the SPD table The packet s destination IP address and optionally destination port source port and protocol type are compared to each entry in the table If a match is found the gateway checks if an SA already exists for this entry If it doesn t the IKE protocol is invoked refer to Section 12 1 1 above and an IPSec SA is established The packet is encrypted and transmitted If a match isn t found the packet is transmitted un encrypted Note An incoming packet whose parameters match one of the entries of the SPD table but received un encrypted is dropped SIP User s Manual 280 Document LTRT 65408 SIP User s Manual 12 Security IPSec specifications include the following m Transport mode only Encapsulation Security Payload ESP only Support for Cipher Block Chaining CBC Supported IPSec SA encryption algorithms DES 3DES and AES Hash types for IPSec SA are SHA1 and MD5 12 1 3 Configuring the IPSec and IKE To enable
393. neContext refer to Section 5 5 13 on page 132 gt 1 Phone Context Table To configure the Phone Context tables take these 6 steps Open the Phone Context Table screen Protocol Management menu gt Manipulation Tables submenu gt Phone Context Table option the Phone Context Table screen appears as shown below Figure 5 15 Phone Context Table Screen Add Phone Context As Prefix Phone Context Index 1 10 Phone Context Table Enable v NPI TON Phone Context 1 Unknown v Unknown v unknown com Private v Level 2 Regional host com E 164 Public v National v na e164 host com v v con 0 Eman m E gt wl nN From the Add Phone Context As Prefix drop down list select Enable to add the received Phone Context parameter as a prefix to outgoing ISDN SETUP Called and Calling numbers if necessary From the Phone Context Index drop down list select the index number SIP User s Manual 96 Document LTRT 65408 SIP User s Manual 5 Web Management 4 Configure the Phone Context table according to Table 5 15 5 Click the Submit button to save your changes 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Several rows
394. nect the FXO MediaPack ports directly to the PBX lines as shown in the diagram below Figure 8 7 MediaPack FXS amp FXO Remote IP Extension FXO MediaPack 10 1 10 2 FXS MediaPack 10 1 10 3 8 12 4 1 Dialing from Remote Extension Phone connected to FXS gt 1 To configure the call take these 6 steps Lift the handset to hear the dial tone coming from PBX as if the phone was connected directly to PBX FXS and FXO MediaPack gateways establish a voice path connection from the phone to the PBX immediately the phone handset is raised Dial the destination number the DTMF digits are sent over IP directly to the PBX All tones heard are generated from the PBX such as Ringback busy or fast busy tones There is one to one mapping between FXS ports and PBX lines The call is disconnected when the phone connected to the FXS goes onhook Version 5 0 243 December 2006 r wl AudioCodes MediaPack 8 12 4 2 Dialing from other PBX line or from PSTN gt To configure the call take these 5 steps 1 Dial the PBX subscriber number the same way as if the user s phone was connected directly to PBX 2 When the PBX rings the FXO MediaPack the ring signal is immediately sent to the phone connected to the FXS MediaPack 3 Once the phone s handset connected to the FXS MediaPack is raised the FXO MediaPack seizes the PBX line and the voice path is established between the phone and the PBX line
395. nel secures the messages of the following phase quick mode in which the IPSec SA properties are negotiated The IKE negotiation is as follows Main mode the main mode creates a secured channel for the quick mode e SA negotiation The peers negotiate their capabilities using two proposals Each proposal includes three parameters Encryption method Authentication protocol and the length of the key created by the DH protocol The key s lifetime is also negotiated in this stage For detailed information on configuring the main mode proposals refer to Section 12 1 3 1 on page 281 e Key exchange DH The DH protocol is used to create a phase 1 key e Authentication The two peers authenticate one another using the pre shared key configured by the parameter IKEPolicySharedKey M Quick mode quick mode negotiation is secured by the phase 1 SA e SA negotiation The peers negotiate their capabilities using a single proposal The proposal includes two parameters Encryption method and Authentication protocol The lifetime is also negotiated in this stage For detailed information on configuring the quick mode proposal refer to the SPD table under Section 12 1 3 2 on page 284 e Key exchange a symmetrical key is created using the negotiated SA IKE specifications include the following Authentication mode pre shared key only m Main mode is supported for IKE Phase 1 E Supported IKE SA encryption algorithms Data Encryption Sta
396. nel Capacity Available Ports FXS Functionality FXS Capabilities Additional Features Polarity Reversal Wink Metering Tones Distinctive Ringing Message Waiting Indication MP 124 24 ports Short or Long Haul Automatic Detection REN2 Up to 15 5 km 50 800 feet using 24 AWG line REN3 Up to 9 km 30 000 feet using 24 AWG line REN5 Up to 5 5 km 18 000 feet using 24 AWG line Note The lines were tested under the following conditions ring voltage greater than 32 Vrms offhook loop current greater than 20 mA all lines ring simultaneously Includes lightning and high voltage protection for outdoor operation Caller ID generation 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 Programmable Line Characteristics Battery feed line current hook thresholds AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains Note For a specific coefficient file contact AudioCodes Programmable ringing signal Up to three cadences and frequency 15 to 200 Hz Over temperature protection for abnormal situations as shorted lines Loop backs for testing and maintenance Immediate or smooth to prevent erroneous ringing 12 16 KHz sinusoidal bursts By frequency 15 100 Hz and cadence patterns DC voltage generation TIA EIA 464 B V23 FSK data Stutter dial tone and DTMF based Voice amp Tone Characteri
397. network connection fails Users can therefore use the Lifeline phone even when the MP 11x is not powered on or not connected to the network On FXS gateways a single Lifeline connected to port 1 via a splitter not supplied is available On combined FXS FXO gateways a splitter isn t required all FXS ports are automatically connected to FXO ports FXS port 0 to FXO port 4 and so forth On FXO gateways a Lifeline isn t available The Lifeline s splitter connects pins 1 and 4 to another source of an FXS port and pins 2 and 3 to the POTS phone Refer to the Lifeline splitter pinouts in Figure 3 8 Figure 3 8 Lifeline Splitter Pinouts and RJ 11 Connector 1 Lifeline Tip 2 Tip 3 Ring 4 Lifeline Ring gt To cable the MP 11x FXS Lifeline take these 3 steps 1 Connect the Lifeline splitter to port 1 on the MP 11x the Lifeline splitter is a special order option 2 Connect the Lifeline phone to Port A on the Lifeline splitter 3 Connect an analog PSTN line to Port B on the Lifeline splitter SIP User s Manual 34 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack gt To cable the combined MP 11x FXS FXO Lifeline take these 2 steps 1 Connect a fax machine modem or phone to each of the FXS ports 2 Connect an analog PSTN line to each of the FXO ports Note The use of the Lifeline on network failure can be disabled using the LifeLineType ini file parameter described
398. nfiguration forces the V 34 fax machine to operate in the slower T 30 mode Version 5 0 221 December 2006 a WH wt AudioCodes MediaPack 8 3 5 Supporting V 152 Implementation The MediaPack gateway 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 gateway 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 When in VBD mode for V 152 implementation support is negotiated between the gateway 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 gomd 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 indic
399. ng Number and optionally a Calling Name P asserted or P preferred headers are used together with the Privacy header If Caller ID is restricted the Privacy id is included Otherwise for allowed Caller ID the Privacy none is used If Caller ID is restricted received from Tel or configured in the gateway the From header is set to lt anonymous anonymous invalid gt Determines the SIP signaling method used to establish and convey a fax session after a fax is detected No Fax 0 No fax negotiation using SIP signaling default T 38 Relay 1 Initiates T 38 fax relay G 711 Transport 2 Initiates fax using the coder G 711 A law p law with adaptations refer to note 1 Fax Fallback 3 Initiates T 38 fax relay If the T 38 negotiation fails the gateway re initiates a fax session using the coder G 711 A law u law with adaptations see note 1 Note 1 Fax adaptations Echo Canceller On Silence Compression Off Echo Canceller Non Linear Processor Mode Off Dynamic Jitter Buffer Minimum Delay 40 Dynamic Jitter Buffer Optimization Factor 13 Note 2 If the gateway initiates a fax session using G 711 option 2 and possibly 3 a gpmd attribute is added to the SDP in the following format For A law a gpmd 0 vbd yes ecan on For p law a gpmd 8 vbd yes ecan on Note 3 When IsFaxUsed is set to 1 2 or 3 the parameter FaxTransportMode is ignored Note 4 When the value of
400. ng standards for detection FXO and generation FXS of Caller ID and detection FXO of MWI when specified signals Bellcore 0 Caller ID and MWI default ETSI 1 Caller ID and MWI NTT 2 British 4 DTMF ETSI 16 Denmark 17 Caller ID and MWI India 18 Brazil 19 Note 1 The Caller ID signals are generated detected between the first and the second rings Note 2 To select the Bellcore Caller ID sub standard use the parameter BellcoreCallerlDTypeOneSubStandard To select the ETSI Caller ID sub standard use the parameter ETS ICallerIDT ypeOneSubStandard Note 3 To select the Bellcore MWI sub standard use the parameter BellcoreVMWITypeOneStandard To select the ETSI MWI sub standard use the parameter ETSIVMWITypeOneStandard Determines a digit pattern which when received from the Tel side indicates a Hook Flash event The valid range is a 25 character string Enable MWI message waiting indication Disable 0 Disabled default Enable 1 MWI service is enabled This parameter is applicable only to FXS gateways Note The MediaPack only supports reception of MWI For detailed information on MWI refer to Section 8 1 6 on page 218 Disable 0 Disable default Enable 1 Enable visual Message Waiting Indication supplies line voltage of approximately 100 VDC to activate the phone s lamp This parameter is applicable only to FXS gateways Disable 0 MWI
401. ng table is used by the gateway to determine IP routing rules It can be used for example to define static routing rules for the OAM and Control networks since a default gateway isn t supported for these networks refer to Section 9 9 1 on page 254 Before sending an IP packet the gateway searches this table for an entry that matches the requested destination host network If such entry is found the gateway sends the packet to the indicated router If no explicit entry is found the packet is sent to the default gateway configured in Network Settings gt IP Settings screen Up to 50 routing entries are available gt To configure the IP Routing table take these 3 steps 1 Open the IP Routing Table screen Advanced Configuration menu gt Network Settings gt Routing Table option the IP Routing Table screen is displayed Figure 1 3 IP Routing Table Screen Routing Table Delete Row Destination IP Address Destination Mask Gateway IP Address TTL Hop Count Network Type Add a new table entry Destination IP Address Destination Mask Gateway IP Address Hop Count Network T D oam z Note All fields should have a value 2 Use the Add a new table entry pane to add a new routing rule Each field in the IP routing table is described in Table 5 39 3 Click the button Add New Entry the new routing rule is added to the IP routing table Table 5 39 IP Routing Table Column Description
402. ng the Charge Codes Table refer to Section 5 5 2 3 1 below Version 5 0 87 December 2006 Ta A wl AudioCodes MediaPack 5 5 2 3 1 Charge Codes Table The Charge Codes table is used to configure the metering tones and their time interval that the FXS gateway generates to the Tel side To associate a charge code to an outgoing Tel to IP call use the Tel to IP Routing table gt 1 To configure the Charge Codes table take these 6 steps Access the Metering Tones screen Protocol Management menu gt Advanced Parameters submenu gt Metering Tones option the Metering Tones screen is displayed Figure 5 11 Open the Charge Codes Table screen by clicking the arrow sign gt to the right of the Charge Codes Table label the Charge Codes Table is displayed Figure 5 12 Charge Codes Table Screen Time Period 2 Time Period 3 Time Period 4 Puses End Time Puls Interval HOSES End Time Puls Interval FOES End Time Puls Interval Pukes O Answer Use the table to define up to 25 different charge codes each charge code is defined in a single row Each code can include from a single and up to four different time periods in a day 24 hours Each time period is composed of e The end in a 24 rounded hour s format of the time period e The time interval between pulses in seconds e The number of pulses sent on answer The first time period always starts at midn
403. nnectOnBrokenConnection 0 the gateway 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 amount of time in 100 msec units an RTP packet isn t received after which a call is disconnected The valid range is 1 to 1000 The default value is 100 10 seconds Note 1 Applicable only if DisconnectOnBrokenConnection 1 Note 2 Currently this feature works only if Silence Suppression is disabled Version 5 0 79 December 2006 Ta fa AudioCodes MediaPack Table 5 8 Advanced Parameters General Parameters continues on pages 78 to 82 Parameter Description Disconnect Call on Silence Yes 1 The MediaPack disconnect calls in which silence occurs in both call Detection directions for more than 120 seconds EnableSilenceDisconnect No 0 Call is not disconnected when silence is detected default The silence duration can be set by the FarEndDisconnectSilencePeriod parameter default 120 Silence Detection Period sec Duration of silence period in seconds prior to call disconnection FarEndDisconnectSilenceP The range is 10 to 28800 8 hours The default is 120 seconds eriod Applicable to gateways that use DSP templates 2 or 3 Silence Detection Method Silence detection method FarEndDisconnectSilenceM None 0 Silence detection option is disabled ethod TelConnectC
404. nsitive the parameter value is not case sensitive except for coder names E The ini file should be ended with one or more carriage returns 6 3 2 The ini File Example Figure 6 2 shows an example of an ini file for the VoIP gateway Figure 6 2 SIP ini File Example Channel Params DJBufMinDelay 75 RTPRedundancyDepth 1 DefaultNumber 101 MaxDigits 3 CoderName g7231 90 Phone of each endpoint EnableSyslog 0 Files CallProgressTonesFilename CPUSA dat FXSLoopCharacteristicsFileName coeff dat SaveConfiguration 1 SIP User s Manual 210 Document LTRT 65408 SIP User s Manual 7 Using BootP DHCP 7 7 1 Using BootP DHCP The MediaPack uses the Bootstrap Protocol BootP and the Dynamic Host Configuration Protocol DHCP to obtain its networking parameters and configuration automatically after it is reset BootP and DHCP are also used to provide the IP address of a TFTP server on the network and files cmp and ini to be loaded into memory DHCP is a communication protocol that automatically assigns IP addresses from a central point BootP is a protocol that enables a device to discover its own IP address Both protocols have been extended to enable the configuration of additional parameters specific to the MediaPack BootP is normally used to initially configure the MediaPack Thereafter BootP is no longer required as all parameters can be stored in the gateway s non vol
405. nsparent to IP applications IPSec and IKE are used in conjunction to provide security for control and management e g SNMP and Web protocols but not for media i e RTP RTCP and T 38 IPSec is responsible for securing the IP traffic This is accomplished by using the Encapsulation Security Payload ESP protocol to encrypt the IP payload illustrated in Figure 12 1 below The IKE protocol is responsible for obtaining the IPSec encryption keys and encryption profile known as IPSec Security Association SA Figure 12 1 IPSec Encryption IPSec doesn t function properly if the gateway s IP address is changed on the fly due to the fact that the crypto hardware can only be configured on reset Therefore reset the gateway after you change its IP address Version 5 0 279 December 2006 7a Ta e wt AudioCodes MediaPack 12 1 1 12 1 2 IKE IKE is used to obtain the Security Associations SA between peers the gateway and the application it s trying to contact The SA contains the encryption keys and profile used by the IPSec to encrypt the IP stream The IKE table lists the IKE peers with which the gateway performs the IKE negotiation up to 20 peers are available The IKE negotiation is separated into two phases main mode and quick mode The main mode employs the Diffie Hellman DH protocol to obtain an encryption key without any prior keys and uses a pre shared key to authenticate the peers The created chan
406. ntained If the gateway is reset from the Web SNMP only a single DHCP sequence containing option 60 is sent If DHCP procedure is used the new gateway IP address allocated by the DHCP server must be detected Note If during operation the IP address of the gateway is changed as a result of a DHCP renewal the gateway is automatically reset gt To detect the gateway s IP address follow one of the procedures below E Starting with Boot version 1 92 the gateway can use a host name in the DHCP request The host name is set to acl_nnnnn where nnnnn stands for the gateway s serial number the serial number is equal to the last 6 digits of the MAC address converted from Hex to decimal If the DHCP server registers this host name to a DNS server the user can access the gateway through a Web browser using a URL of http acl_ lt serial number gt instead of using the gateway s IP address For example if the gateway s MAC address is 00908f010280 the DNS name is acl_66176 m After physically resetting the gateway its IP address is displayed in the Client Info column in the BootP TFTP configuration utility refer to Figure C 1 on page 351 m Use a serial communication software refer to Section 4 2 4 on page 44 E Contact your System Administrator SIP User s Manual 212 Document LTRT 65408 SIP User s Manual 7 Using BootP DHCP 7 3 7 3 1 Using BootP Upgrading the MediaPack When up
407. nto the MediaPack Table 5 64 below describes the ini file parameters that are associated with the configuration files Table 5 64 Configuration Files ini File Parameters ini File Parameter Name CallProgressTonesFileName FXSLoopCharacteristicsFileName PrerecordedTonesFileName UserlInfoFileName SaveConfiguration Description The name and path of the file containing the Call Progress Tones definition The name and path of the file providing the FXS line characteristic parameters The name and path of the file containing the Prerecorded Tones The name and path of the file containing the User Information data Determines if the gateway s configuration parameters and files is saved to flash non volatile memory 0 Configuration isn t saved to flash memory 1 Configuration is saved to flash memory default Version 5 0 203 December 2006 r wi AudioCodes MediaPack 5 10 5 10 1 Maintenance The Maintenance menu is used for the following operations E Locking and unlocking the gateway refer to Section 5 10 1 on page 204 H Saving the gateway s configuration refer to Section 5 10 2 on page 205 E Resetting the gateway refer to Section 5 10 3 on page 206 Locking and Unlocking the Gateway The Lock and Unlock options allow you to lock the gateway so that it does not accept any new incoming calls This is beneficial when for example you are uploading new software files to th
408. nto two similar but slightly different types that reflect the organization of the tree structure Discrete MIB Objects Contain one precise piece of management data These objects are often distinguished from Table items below by adding a 0 dot zero extension to their names The operator must merely know the name of the object and no other information Table MIB Objects Contain multiple sections of management data These objects are distinguished from Discrete items above by requiring a dot extension to their names that uniquely distinguishes the particular value being referenced The dot extension is the instance number of an SNMP object For Discrete objects this instance number is zero For Table objects this instance number is the index into the SNMP table SNMP tables are special types of SNMP objects which allow parallel arrays of information to be supported Tables are distinguished from scalar objects so that tables can grow without bounds For example SNMP defines the ifDescr object as a standard SNMP object that indicates the text description of each interface supported by a particular device Since network devices can be configured with more than one interface this object can only be represented as an array By convention SNMP objects are always grouped in an Entry directory within an object with a Table suffix The ifDescr object described above resides in the ifEntry directory con
409. nu gt Endpoint Settings gt Authentication option the Authentication screen is displayed Figure 5 27 Authentication Screen Authentication ET ELG Password 3 In the User Name and Password fields for a port enter the username and password combination respectively 4 Repeat Step 4 for each port 5 Click the Submit button to save your changes 6 To save the changes refer to Section 5 10 2 on page 205 Table 5 27 Authentication ini File Parameter Parameter Name in ini File Parameter Format Authentication_x Authentication_ lt Port gt lt Username gt lt Password gt For example Authentication_0 david 14325 Authentication_1 Alex 18552 Note Using the sign enables the user to omit either the username or the password For instance Authentication_5 152 In this case endpoint 5 s phone number is used instead of username Version 5 0 119 December 2006 7a Ta wt AudioCodes MediaPack 5 5 9 2 Automatic Dialing Use the Automatic Dialing Table to define telephone numbers that are automatically dialed when a specific port is used gt To configure the Automatic Dialing table take these 6 steps 1 Open the Automatic Dialing screen Protocol Management menu gt Endpoint Settings submenu gt Automatic Dialing option the Automatic Dialing screen is displayed Figure 5
410. nual 4 Getting Started eo Y To assign an IP address using BootP take these 3 steps Open the BootP application supplied with the MediaPack software package Add client configuration for the MediaPack refer to Section C 11 1 on page 355 Use the reset button to physically reset the gateway causing it to use BootP the MediaPack changes its network parameters to the values provided by the BootP 4 2 3 Assigning an IP Address Using the Voice Menu Guidance Initial configuration of the gateway can be performed using a standard touch tone telephone connected to one of the FXS analog ports The voice menu can also be used to query and modify basic configuration parameters gt To assign an IP address using the voice menu guidance take these 7 steps Connect a telephone to one of the FXS ports Lift the handset and dial 12345 three stars followed by the digits 1 2 3 4 5 Wait for the configuration menu voice prompt to be played To change the IP address press 1 followed by the pound key e The current IP address of the gateway is played Press to change it e Dial the new IP address use the star key instead of dots e g 192 168 0 4 and press to finish Review the new IP address and press 1 to save it To change the subnet mask press 2 followed by the key e The current subnet mask of the gateway is played Press to change it e Dial the new subnet mask e g 255 255 0 0 and pres
411. o Version description Call Progress Tones Fie CPT version version2 V Use dBm units for Tone Levels Make File 3 Click the Select File button that is in the Call Progress Tone File box 4 Navigate to the folder that contains the CPT ini file you want to convert 5 Click the ini file and click the Open button the name and path of both the ini file and the output dat file appears in the fields below the Select File button 6 Enter the Vendor Name Version Number and Version Description in the corresponding required fields under the User Data section e The maximum length of the Vendor field is 256 characters e The format of the Version field is composed of two integers separated by a period e g 1 2 23 4 5 22 e The maximum length of the Version Description field is 256 characters 7 The default value of the CPT Version drop down list is Version 3 Do one of the following e Ifthe software version you are using is prior to version 4 4 select Version 1 to maintain backward compatibility e If the software version you are using is 4 4 select Version 2 e Otherwise leave the value at its default 8 Check the Use dBm units for Tone Levels check box Note that the levels of the Call Progress Tones in the CPT file must be in dBm units 9 Click the Make File button you re prompted that the operation conversion was successful 10 Close the application SIP Us
412. o A Record no NAPTR or SRV queries are performed If set to SRV and the Proxy IP address parameter contains a domain name without port definition e g ProxyIP domain com an SRV query is performed The SRV query returns up to four Proxy host names and their weights The gateway then performs DNS A record queries for each Proxy host name according to the received weights to locate up to four Proxy IP addresses Therefore if the first SRV query returns two domain names and the A record queries return two IP addresses each no more searches are performed If set to NAPTR 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 IP address parameter contains a domain name with port definition e g ProxylP domain com 5080 the gateway performs a regular DNS A record query Note When enabled NAPTR SRV queries are used to discover Proxy servers even if the parameter DNSQueryType is disabled aan Proxy SRV This parameter is now obsolete Please use the parameter ProxyDNSQueryType ueries EnableProxySRVQuery Redundancy Mode Parking 0 Gateway continues working with the last active Proxy until the next ProxyRedundancyMod failure default e Homing 1 Gateway always tries to work with the primary Proxy server switches back to the main Proxy wheneve
413. o Section 5 10 2 on page 205 Version 5 0 173 December 2006 Ta WH wt AudioCodes MediaPack Table 5 50 Security Settings General Security Settings Parameters continues on pages 174 to 175 Parameter Description Secured Web Connection Determines the protocol types used to access the Embedded Web Server HTTPSOnly HTTP and HTTPS 0 default HTTPS only 1 unencrypted HTTP packets are blocked HTTP Authentication Mode Determines the authentication mode for the Embedded Web Server WebAuthMode Basic 0 Basic authentication clear text is used default Digest When Possible 1 Digest authentication MD5 is used Basic if HTTPS Digest if HTTP 2 Digest authentication MD5 is used for HTTP and basic authentication is used for HTTPS Note that when RADIUS login is enabled WebRADIUSLogin 1 basic authentication is forced Voice Menu Password Password for the voice menu used for configuration and status To activate VoiceMenuPassword the menu connect an analog telephone and dial three stars followed by the password The default value is 12345 For detailed information on the voice menu refer to Section 4 2 3 on page 43 RADIUS General Settings Enable RADIUS Access Control Enables disables the RADIUS application EnableRADIUS Disable 0 RADIUS application is disabled default Enable 1 RADIUS application is enabled Note In the current version RADIUS is used only for HTTP a
414. o Table 9 1 on page 255 in which it operates m Two separate IP addresses Dual IP mode one for a specific traffic type and the other for a combination of two traffic types In Dual IP mode the gateway is assigned two IP addresses for the different traffic types One IP address is assigned to a combination of two traffic types Media and Control OAM and Control or OAM and Media while the other IP address is assigned to whichever traffic type that is not included in this combination For example a typical scenario using this mode would include one IP address assigned for Control and OAM and another IP address assigned for Media For detailed information on integrating the MediaPack into a VLAN and multiple IPs network refer to Section 9 9 3 on page 256 For detailed information on configuring the multiple IP parameters refer to Section 5 6 1 1 on page 138 A default gateway is supported only for the Media traffic type for the other two use the IP Routing table The IP address and subnet mask used in the Single IP Network mode are carried over to the OAM traffic type in the Multiple IP 9 9 2 Network mode IEEE 802 1p Q VLANs and Priority The Virtual Local Area Network VLAN mechanism enables the MediaPack to be integrated into a VLAN aware environment that includes switches routers and endpoints When in VLAN enabled mode each packet is tagged with values that specify its priority class of service IEEE 802 1p
415. o access the gateway s Web and Telnet interfaces Access from an undefined IP address is denied This security feature is inactive the gateway can be accessed from any IP address when the table is empty gt To manage the Web amp Telnet access list take these 4 steps 1 Open the Web amp Telnet Access List screen Advanced Configuration menu gt Security Settings gt Web amp Telnet Access List option the Web amp Telnet Access List screen is displayed Figure 5 48 Web amp Telnet Access List Screen Web amp Telnet Access List Delete Row Authorized IP Address Note Delete all rows to allow access from any IP address Add a new IP address authorized to connect to the device s web and telnet interfaces New Authorized IP Address 2 To add a new authorized IP address in the New Authorized IP Address field enter the required IP address refer to Note 1 below and click Add New Address the IP address you entered is added as a new entry to the Web amp Telnet Access List table 3 To delete authorized IP addresses check the Delete Row checkbox in the rows of the IP addresses you want to delete refer to Note 2 below and click the button Delete Selected Addresses the IP addresses are removed from the table and can no longer access the Web amp Telnet interfaces 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 The first authorized IP
416. o different IP addresses can be assigned to the same hostname If the hostname isn t found in this table the gateway communicates with an external DNS server Assigning two IP addresses to hostname can be used for alternative routing using the Tel to IP Routing table gt 1 To configure the internal DNS table take these 7 steps Open the Internal DNS Table screen Protocol Management menu gt Routing Tables submenu gt Internal DNS Table option the Internal DNS Table screen is displayed Figure 5 19 Internal DNS Table Screen Internal DNS Table _ Domain Name _First IP Address _Second IP Address DomainNake com 10 8 21 4 10 13 2 95 2 3 4 5 E 7 8 9 _ 2 In the Domain Name field enter the hostname to be translated You can enter a string up to 31 characters long 3 In the First IP Address field enter the first IP address that the hostname is translated to 4 In the Second IP Address field enter the second IP address that the hostname is translated to 5 Repeat steps 2 to 4 for each Internal DNS Table entry 6 Click the Submit button to save your changes 7 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 19 Internal DNS ini File Parameter Parameter Name in ini File Parameter Format DNS2IP DNS2IP lt
417. oCodes MediaPack E 2 6 The Call Progress Tones ini File After the Call Progress Tones detection is complete a text file named call_progress_tones ini is created in the same directory as the directory in which the CPTWizard exe is located This file contains E Information about each tone that was recorded and analyzed by the wizard This information includes frequencies and cadence on off times and is required for using this file with the TrunkPack Downloadable Conversion utility Figure E 10 Call Progress Tone Properties CALL PROGRESS TONE 1 Tone Type 1 Low Freq Hz 350 High Freq Hz 440 Low Freq Level dBm 0 High Freq Level dBm 0 First Signal On Time 10msec 0 First Signal Off Time 10msec 0 Second Signal On Time 10msec 0 Second Signal Off Time 10msec 0 m Information related to possible matches of each tone with the CPTWizard s internal database of well known tones This information is specified as comments in the file and is ignored by the TrunkPack Downloadable Conversion utility Figure E 11 Call Progress Tone Database Matches Recorded tone Busy Tone automatic configuration Matches PBX name ITU Anguilla Tone name Busy tone Matches PBX name ITU Antigua and Barbuda Tone name Busy tone Matches PBX name ITU Barbados Tone name Busy tone Matches PBX name ITU Bermuda Tone name Busy tone Matches PBX name ITU British Virgin Islan Tone name Bus
418. oardCallResourcesAlarm acBoardControllerFailureAlarm acFeatureKeyError acBoardOverloadAlarm acActiveAlarmTableOverflow acKeepAlive acNATTraversalAlarm acEnhancedBITStatus acBoardEthernetLinkAlarm acPerformanceMonitoringThresholdCrossi ng acHTTPDownloadResult Description Sent whenever a fatal device error occurs Sent when a device s settings are illegal The trap contains a message describing the illegality of the setting Note Not applicable to IPM 260 Sent when a board exceeds its temperature limits Sent after the device is reset Sent after the device is successfully restored and initialized following reset Sent when Graceful Shutdown commences and ends Sent if the operational state of the node changes to disabled Cleared when the operational state of the node changes to enabled Sent when no free channels are available Sent when the Gatekeeper Proxy is not found or registration failed Internal routing table can be used for routing Intended to relay Feature Key errors etc will be supported in the next applicable release Sent when an overload in one or more of the system s components occurs Sent to indicate that an active alarm could not be entered into the Active Alarm table because the table was full Part of the NAT traversal mechanism If the STUN application detects a NAT this trap is sent on regular time laps 9 10 of the acSysSTUNBindingLifeTime object The Additionalln
419. ode Send Digit Pattern on Connect CDR and Debug CDR Server IP Address CDRSyslogServerIP CDR Report Level CDRReporiLevel Debug Level GwDebugLevel Misc Parameters Progress Indicator to IP ProgressIndicator2IP Packets Count 1 According to packet count Voice Energy Detectors 2 According to energy and voice detectors default All 3 According to packet count and energy voice detectors Defines a digit pattern that is sent to the Tel side after 200 OK is received from the IP side The digit pattern is a predefined DTMF sequence that is used to indicate an answer signal e g for billing purposes Applicable only to FXS gateways The valid range is 1 to 8 characters Defines the destination IP address for CDR logs The default value is a null string that causes the CDR messages to be sent with all Syslog messages Note The CDR messages are sent to UDP port 514 default Syslog port None 0 Call Detail Recording CDR information isn t sent to the Syslog server default End Call 1 CDR information is sent to the Syslog server at end of each Call Start amp End Call 2 CDR information is sent to the Syslog server at the start and at the end of each Call The CDR Syslog message complies with RFC 3161 and is identified by Facility 17 local1 and Severity 6 Informational Syslog logging level One of the following debug levels can be selected 0 0 Debug is disabled d
420. ode set e RxDTMFOption 0 Protocol Management gt Protocol Definition gt DTMF amp Dialing gt Declare RFC 2833 in SDP No e TxDTMFOption 1 1 to 5 DTMF Option INFO Nortel Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Mute 2 Using INFO message according to Cisco s mode In this mode DTMF digits are carried to the remote side within INFO messages To enable this mode set e RxDTMFOption 0 Declare RFC 2833 in SDP No e TxDTMFOption 3 1 to 5 DTMF Option INFO Cisco Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Mute SIP User s Manual 218 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 3 Using NOTIFY messages according to lt draft mahy sipping signaled digits 01 txt gt In this mode DTMF digits are carried to the remote side using NOTIFY messages To enable this mode set e RxDTMFOption 0 Declare RFC 2833 in SDP No e TxDTMFOption 2 1 to 5 DTMF Option NOTIFY Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Mute 4 Using RFC 2833 relay with Payload type negotiation In this mode DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard To enable this mode
421. ol 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 The protocol field also accepts the abbreviated strings SIP MGCP MEGACO 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 the Packet Size field relates to the overall reassembled packet size not to the size of each fragment Expected traffic rate bytes per second Tolerance of traffic rate limit number of bytes Action upon match allow or block A read only field that provides the number of packets accepted rejected by a specific rule 5 6 5 4 Configuring the Certificates Use the Certificates screen to replace the server refer to Section 12 2 4 on page 290 and client refer to Section 12 2 5 on page 292 certificates and to update the private key HTTPSPkeyFileName described in Table 5 55 on page 182 SIP User s Manual 172 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 5 5 Configuring the General Security Settings gt To configure the General Security Settings parameters take these 4 steps 1 Open the General Security Settings screen Advanced Configuration menu gt Security
422. ollowed by repeated ringing of 1 sec on and 3 sec off 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 10msec 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 Version 5 0 327 December 2006 7a Ta r wt AudioCodes MediaPack 15 1 2 Format of the Distinctive Ringing Section in the ini File Distinctive Ringing is only applicable to MediaPack FXS gateways Using the distinctive ringing section of this configuration file the user can create up to 16 distinctive ringing patterns Each ringing pattern configures the ringing tone frequency and up to 4 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 Hz to 200 Hz with a 5 Hz resolution Each of the ringing pattern cadences is specified by the following parameters 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 Other
423. om the Packetization Time drop down list select the packetization time in msec for the coder you selected 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 coder you selected 5 In the Payload Type field if the payload type for the coder is dynamic enter a value from 0O to 120 payload types of well known coders cannot be modified The payload type identifies the format of the RTP payload 6 From the Silence Suppression drop down list enable or disable the silence suppression option for the coder you selected 7 Repeat steps 2 through 6 for the second to fifth coders optional 8 Click the Submit button to save your changes 9 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Each coder can appear only once If not specified the ptime gets a default value The ptime specifies packetization time the gateway expects to receive The gateway always uses the ptime requested by the remote side for sending RTP packets Only the ptime of the first coder in the list is declared in INVITE 200 OK SDP even if multiple coders are defined If payload type is not specified a default is used For G 729 it s also possible to select silence suppression without adaptations If coder G 729 is selected and silence suppression disabled the
424. on 4 The protective earth terminal on the back of the MP 124 must be permanently connected to protective earth Safety Notices The safety status of each port on the gateway is declared and detailed in the table below TNV 3 Circuit whose normal operating voltages exceeds the limits for an SELV circuit under normal operating conditions and on which over voltages from Telecommunication Networks are possible SELV Safety extra low voltage circuit Telecommunication Safety Ports Safety Status Ethernet 100 Base TX SELV FXS TNV 3 FXO TNV 3 SIP User s Manual 388 Document LTRT 65408 SIP User s Manual H Regulatory Information Industry Canada Notice This equipment meets the applicable Industry Canada Terminal Equipment technical specifications This is confirmed by the registration numbers The abbreviation IC before the registration number signifies that registration was performed based on a declaration of conformity indicating that Industry Canada technical specifications were met It does not imply that Industry Canada approved the equipment FXO Ports The Ringer Equivalence Number REN for this terminal is 0 05 The REN assigned to each terminal equipment provides an indication of the maximum number of terminals allowed to be connected to a telephone interface The termination on an interface may consist of any combination of devices subject only to the requirement that the sum of Ringer Equ
425. onds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call If the gateway receives a 415 Unsupported Media response it notifies the User with a Reorder Tone The gateway generates this response in case of SDP mismatch The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call If the gateway receives a 480 Temporarily Unavailable response it notifies the User with a Reorder Tone This response is issued if there is no response from remote The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message be
426. onf FreeRADIUS Client Configuration clients conf client configuration directives client 10 31 4 47 secret FutureRADIUS shortname tp1610_master_tpm SIP User s Manual 294 Document LTRT 65408 SIP User s Manual 12 Security 2 If access levels are required set up a VSA dictionary for the RADIUS server and select an attribute ID that represents each user s access level The following example shows a dictionary file for FreeRADIUS that defines the attribute ACL Auth Level with ID 35 Figure 12 13 Example of a Dictionary File for FreeRADIUS FreeRADIUS Client Configuration AudioCodes VSA dictionary VENDOR AudioCodes 5003 ATTRIBUTE ACL Auth Level 35 integer AudioCodes VALUE ACL Auth Level ACL Auth UserLevel 50 VALUE ACL Auth Level ACL Auth AdminLevel 100 VALUE ACL Auth Level ACL Auth SecurityAdminlLevel 200 3 In the RADIUS server define the list of users authorized to use the gateway using one of the password authentication methods supported by the server implementation The following example shows a user configuration file for FreeRADIUS using a plain text password Figure 12 14 Example of a User Configuration File for FreeRADIUS Using a Plain Text Password users local user configuration database john Auth Type Local User Password qwerty Service Type Login User ACL Auth Level ACL Auth SecurityAdminLevel Auth Type Local User Password 123456 Servi
427. onfigure the attributes of up to five SNMP managers gt To configure the SNMP Managers Table take these 5 steps 1 Access the Management Settings screen Advanced Configuration menu gt Management Settings the Management Settings screen is displayed Figure 5 51 2 Open the SNMP Managers Table screen by clicking the arrow sign gt to the right of the SNMP Managers Table label the SNMP Managers Table screen is displayed Figure 5 52 3 Configure the SNMP Managers parameters according to Table 5 52 as Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Figure 5 52 SNMP Managers Table Screen SNMP Managers Table IP Address Trap Port Trap Enable SNMP Manager 1 0 0 0 0 Enable gt SNMP Manager 2 0 0 0 0 Enable gt SNMP Manager SNMP Manager m E 3 pooo fiee Enae 4 pooo fee Erbe H SIP User s Manual 178 Document LTRT 65408 SIP User s Manual 5 Web Management Note If you clear a checkbox and click Submit all settings in the same row revert to their defaults Table 5 52 SNMP Managers Table Parameters Parameter Checkbox SNMPManagerlsUsed_x IP Address SNMPManagerTablelP_x Trap Port SNMPManagerTrapPort_x Trap Enable SNMPManagerTrapSending Enable_x Description Up to five param
428. option KeyCLIRDeact After the sequence is pressed a confirmation tone is heard Hotline Note that the destination phone number and the auto dial status can be viewed in the Automatic Dialing table refer to Section 5 5 9 2 on page 120 Activate Keypad sequence that activates the delayed hotline option KeyHotLine To activate the delayed hotline option from the telephone Dial the preconfigured sequence number on the keypad a dial tone is heard Dial the telephone number to which the phone automatically dials after a configurable delay terminate the number with a confirmation tone is heard Deactivate Keypad sequence that deactivates the delayed hotline option KeyHotLineDeact After the sequence is pressed a confirmation tone is heard SIP User s Manual 90 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 3 Configuring the Number Manipulation Tables The VoIP gateway provides four Number Manipulation tables for incoming and outgoing calls These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly The Manipulation Tables are m Destination Phone Number Manipulation Table for IP gt Tel calls Destination Phone Number Manipulation Table for Tel gt IP call E Source Phone Number Manipulation Table for IP gt Tel calls E Source Phone Number Manipulation Table for Tel gt IP calls Number manipulation can occur either before or after a routing decis
429. or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Not applicable set to f Not applicable set to g Source number prefix h Not applicable set to i Source IP address obtained from the Contact header in the INVITE message The b to d manipulation rules are applied if the called and calling numbers match the a g and i conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example NumberMapIP2Tel 01 2 972 034 10 13 77 8 NumberMapIP2Tel 03 2 667 22 Note The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The wildcard represents any number between 0 and 255 e g 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 SourceNumberMapTel2IP a b c d e f g h a Source number prefix b Number of stripped digits from the left or if in brackets are used from right A combination of both options is allowed c String to add as prefix or if in brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Number Plan used in RPID header f Number Type used in RPID header g Destin
430. orded definitions as they might be more accurate For full operability of the MediaPack FXO gateway it may be necessary to edit this file and add more Call Progress Tone definitions Sample Call Progress Tones ini files are available in the release package When the CPT ini file is complete use the TrunkPack Downloadable Conversion utility to create a loadable CPT dat file After loading this file to the gateway repeat the automatic detection procedure discussed above and verify that the gateway detects all four Call Progress Tones correctly Version 5 0 373 December 2006 a WH wt AudioCodes MediaPack E 2 7 Adding a Reorder Tone to the CPT File The following procedure describes how to add a Reorder tone that a PBX generates to indicate a disconnected call to the CPT file gt 1 10 11 To add a Reorder tone to the CPT file take these 11 steps Make a call using G 711 between the MP FXO which is connected to the PBX and a remote entity in the IP network Capture the call using a network sniffer such as Wireshark Disconnect the call from the PBX side and then wait approximately 30 seconds before stopping the Wireshark recording In the network trace locate the RTP stream sent from the FXO Save the RTP payload on your PC as a pcm file by clicking Save Payload Statistics menu gt RTP gt Stream Analysis Note ensure that you select the forward option Open the pcm fil
431. ording to the table below 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 31 Call Forward Table continues on pages 124 to 125 Parameter Description Forward Type Deactivate 0 Don t forward incoming calls default On Busy 1 Forward incoming calls when the gateway port is busy Unconditional 2 Forward any incoming call to the Phone number specified No Answer 3 Forward incoming calls that are not answered with the time specified in the Time for No Reply Forward field On Busy or No Answer 4 Forward incoming calls when the port is busy or when calls are not answered after a configurable period of time Do Not Disturb 5 Immediately reject incoming calls Forward to Phone Number Enter the telephone number or URI number IP address to which the call is forwarded Note If this field only contains telephone number and Proxy isn t used the forward to phone number must be specified in the Tel to IP Routing table of the forwarding gateway SIP User s Manual 124 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 31 Call Forward Table continues on pages 124 to 125 Parameter Time for No Reply Forward Parameter Name in ini File Fwdinfo_x Description If you have set the Forward Type for this port to no reply enter the number of
432. ort 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 This parameter cannot be changed on the fly and requires a gateway reset For detailed information on STUN refer to Section 9 2 1 on page 248 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 Instead of using this parameter you can define the STUN server s domain name using the ini file parameter StunServerDomainName 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 For detailed information on configuring the NFS table refer to Section 5 6 1 3 on page 143 SIP User s Manual 142 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 1 3 Configuring the NFS Settings Network File System NFS enables the MediaPack to access a remote server s shared files and directories and to handle them as if they re located locally A file system the NFS is independent of machine types OSs and network architectures Up to five different NFS file systems can be configured NFS is utilized by the MediaPack to load the cmp ini and configuration files via the Automatic Update mechanism refer to Section 10 3 on page 263 Note that an NFS file server can share mul
433. ote that it s case sensitive to the IP address e g http 10 1 229 17 AdminPage 3 Click Image Load to Device the Image Download screen is displayed shown in Figure 10 7 Figure 10 7 Image Download Screen Send Logo Image file fram your computer to the device Browse MESSE Send Background Image file from your computer to the device Browse Send File Logo width 339 Set Logo Width This button restores the default images Important Use the Save Configuration Link in order to save loaded images to flash memory 4 Click the Browse button in the Send Logo Image File from your computer to the device box Navigate to the folder that contains the logo image file you want to load 5 Click the Send File button the file is sent to the device When loading is complete the screen is automatically refreshed and the new logo image is displayed 6 Note the appearance of the logo If you want to modify the width of the logo the default width is 339 pixels in the Logo Width field enter the new width in pixels and click the Set Logo Width button 7 To save the image to flash memory so it is available after a power fail refer to Section 5 10 2 on page 205 The new logo appears on all Web Interface screens Tip If you encounter any problem during the loading of the files or you want to restore the default images click the Restore Default Images button SIP User
434. outing table enables the gateway to only accept calls originating in the Proxy server and rejects all other calls Filter Calls to IP Don t Filter 0 Disabled default FilterCalls2IP Filter 1 Enabled If the filter calls to IP feature is enabled then when a Proxy is used the gateway first checks the Tel gt IP routing table before making a call through the Proxy If the number is not allowed number isn t listed or a Call Restriction routing rule IP 0 0 0 0 is applied the call is released Enable Digit Delivery to IP Disable 0 Disabled default EnableDigitDelivery2IP Enable 1 Enable digit delivery to IP The digit delivery feature enables sending of DTMF digits to the destination IP address after the Tel gt IP call was answered To enable this feature modify the called number to include at least one p character The gateway uses the digits before the p character in the initial INVITE message After the call was answered the gateway waits for the required time of p 1 5 seconds and then sends the rest of the DTMF digits using the method chosen in band out of band Note The called number can include several p characters 1 5 seconds pause For example the called number can be as follows 1001pp699 8888p9p300 Enable Digit Delivery to Tel Disable 0 Disabled default EnableDigitDelivery Enable 1 Enable Digit Delivery feature for MediaPack FXO amp FXS The digit de
435. pAlive trap is instigated it does not stop The manager can see the NAT type in the MIB audioCodes 5003 acProducts 9 acBoardMibs 10 acSystem 10 acSystemStatus 2 acSysNetwork 6 acSysNAT 2 acSysNATType 1 The manger also has access to the STUN client configuration audioCodes 5003 acProducts 9 acBoardMibs 10 acSystem 10 acSystemConfigura tion 1 acSysNetworkConfig 3 acSysNATTraversal 6 acSysSTUN 21 acNATTraversalAlarm When the NAT is placed in front a device that is identified as a symmetric NAT this alarm is raised It is cleared when a non symmetric NAT or no NAT replace the symmetric one SIP User s Manual 322 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management 14 11 SNMP Administrative State Control 14 11 1 Node Maintenance Node maintenance for the MediaPack is provided by an SNMP interface The acBoardMIB provides two parameters for graceful and forced shutdowns of the MediaPack E acgwAdminState E acgwAdminStateLockControl The acgwAdminState is used either to request set a shutdown 0 undo shutdown 2 or to view get the gateway condition 0 locked 1 shutting down 2 unlocked The acgwAdminStateLockControl is used to set a time limit in seconds for the shutdown where 0 means shutdown immediately forced 1 means no time limit graceful and x where x gt 0 indicates a time limit in seconds timed limit is considered a graceful shutdown Note The acgwAdminStateLoc
436. pating in an IKE exchange must have a prior out of band knowledge of the common key see IKEPolicySharedKey parameter Note 2 For RSA signature based authentication peers must be loaded with a certificate signed by a common CA For additional information on certificates refer to Section 12 2 4 on page 290 Determines the time in seconds the SA negotiated in the first IKE session main mode is valid After the time expires the SA is re negotiated The default value is 28800 8 hours Determines the lifetime in kilobytes the SA negotiated in the first IKE session main mode is valid After this size is reached the SA is re negotiated The default value is 0 this parameter is ignored The lifetime parameters IKEPolicyLifelnSec and IKEPolicyLifelnKB determine the duration the SA created in the main mode phase is valid When the lifetime of the SA expires it is automatically renewed by performing the IKE first phase negotiations To refrain from a situation where the SA expires a new SA is being negotiated while the old one is still valid As soon as the new SA is created it replaces the old one This procedure occurs whenever an SA is about to expire If no IKE methods are defined Encryption Authentication DH Group the default settings shown in Table 12 2 below are applied Proposal 0 Proposal 1 Proposal 2 Proposal 3 Table 12 2 Default IKE First Phase Proposals Encryption Authentication DH Group 3DES SHA1 1
437. points If an incoming IP call is designated to a busy port the called party hears call waiting tone several configurable short 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 Called party can accept the new call using hook flash and can toggle between the two calls To enable Call Waiting E Set EnableCallWaiting 1 E Set EnableHold 1 E 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 Table 5 35 To configure the Call Waiting indication tone cadence modify the following parameters NumberOfWaitingIndications WaitingBeepDuration and TimeBetweenWaitingIndications To configure a delay interval before a Call Waiting Indication is played to the currently busy port use the parameter TimeBeforeWaitingIndication This enables the caller to hang up before disturbing the called party with Call Waiting Indications Applicable only to FXS gateways Both the calling and the called sides are supported by FXS gateways the FXO gateways support only the calling side To indicate Call Waiting the gateway sends a 182 call queued response The gateway identifies a Waiting Call when a 182 call queued response is received Version 5 0
438. ponses the GW releases the call sending appropriate release cause 604 Does Not Exist Anywhere to PSTN side 606 Not Acceptable Version 5 0 347 December 2006 7 T w Reader s Notes SIP User s Manual 348 Document LT RT 65408 SIP User s Manual C BootP TFTP Configuration Utility C BootP TFTP Configuration Utility The BootP TFTP utility enables you to easily configure and provision our boards and media gateways Similar to third party BootP TFTP utilities which are also supported but with added functionality our BootP TFTP utility can be installed on Windows 98 or Windows NT 2000 XP The BootP TFTP utility enables remote reset of the device to trigger the initialization procedure BootP and TFTP It contains BootP and TFTP utilities with specific adaptations to our requirements C 1 When to Use the BootP TFTP The BootP TFTP utility can be used with the device as an alternative means of initializing the gateways Initialization provides a gateway with an IP address subnet mask and the default gateway IP address The tool also loads default software ini and other configuration files BootP Tool can also be used to restore a gateway to its initial configuration such as in the following instances m The IP address of the gateway is not known m The Web browser has been inadvertently turned off m The Web browser password has been forgotten The gateway has encountered a fault that cannot be r
439. ppear in the Boot file field To use both file names use the separator without blank spaces between the xxx cmp and the yyy ini files e g ram cmp SIPqw ini m iniFile This field specifies the configuration ini file that the gateway uses to program its various settings Enter the name of the file that is loaded by the TFTP utility to the VoIP gateway after it receives the BootReply message The actual ini file is located in the TFTP utility directory that is specified in the BootP Preferences window SIP User s Manual 358 Document LTRT 65408 SIP User s Manual C BootP TFTP Configuration Utility C 11 6 Switch fb em Using Command Line Switches You can add command line switches in the field Boot File gt To use a Command Line Switch take these 4 steps 1 In the field Boot File leave the file name defined in the field as it is e g ramMxxx cmp 2 Place your cursor after cmp 3 Press the space bar 4 Type in the switch you require Example ramxxx cmp fb to burn flash memory ramxxx cmp fb em 4 to burn flash memory and for Ethernet Mode 4 auto negotiate Table C 1 lists and describes the switches that are available Table C 1 Command Line Switch Descriptions Description Burn ram cmp in flash only for cmp files Use this switch to set Ethernet mode 0 10 Base T half duplex 1 10 Base T full duplex 2 100 Base TX half duplex 3 100 Base TX full duplex
440. ption 2 2 2 2 1 MP 124 Front Panel MP 124 Physical Description Figure 2 3 illustrates the front layout of the MP 124 Table 2 3 describes the Reset button located on the front panel Table 2 4 lists and describes the front panel LEDs Figure 2 3 MP 124 Front Panel Type Reset button Table 2 3 Front Panel Buttons on the MP 124 Function Resets the MP 124 Comment Press the reset button with a paper clip or any other similar pointed object until the gateway is reset Restores the MP 124 parameters to their factory default values Refer to Section 10 1 on page 261 Table 2 4 Indicator LEDs on the MP 124 Front Panel Label Type Color State Function Ready Device Status Green On Device Powered self test OK Orange Blinking Software Loading Initialization Red o On Malfunction LAN Ethernet Link Gee On Valid 10 100 Base TX Ethernet connection Status Red On Malfunction Conirol Control Link Green Blinking Sending and receiving SIP messages Blank No traffic Data Packet Status Z Blinking Transmitting RTP Real Time Transport Protocol Packets DRE Binkng Receiving RTP Packets Blank No traffic Channels Telephone On Offhook Ringing for FXS Phone Port Interface FXO Line Seize Ringing State for Line Port Blinking There s an incoming call before answering On Line Malfunction Blank Normal Version 5 0 27 December 2006 r wi AudioCodes MediaPack 2 2 2 MP 124 Rear Pan
441. puter that is running the BootP Tool You can set this value to any whole digit Once set the BootP Tool can send that number of BootReply messages to the destination immediately after you send a remote reset to a VoIP gateway at a valid IP address This enables the replies to get through to the VoIP gateway even if the BootRequest is blocked by the firewall To turn off this feature set the Number of Timed Replies 0 C 10 2 TFTP Preferences Enabled To enable the TFTP functionality of the BootP Tool check the box beside this heading If you want to use another TFTP application other than the one included with the BootP Tool unselect the box On Interface This pull down menu displays all network interfaces currently available on the computer Select the interface that you want to use for the TFTP Normally there is only one choice Directory This option is enabled only when the TFTP is enabled Use this parameter to specify the folder that contains the files for the TFTP utility to manage cmp ini Call Progress Tones etc Boot File Mask Boot File Mask specifies the file extension used by the TFTP utility for the boot file that is included in the BootReply message This is the file that contains VoIP gateway software and normally appears as cmp ini File Mask ini File mask specifies the file extension used by the TFTP utility for the configuration file that is included in the BootReply message This is the file that contains Vo
442. r 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 gateway 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 Both FXO detection and FXS generation are supported Telephone to IP Calls The FXO gateway provides the following FXO operating modes for Tel to IP calls m Automatic Dialing E Collecting Digits Mode E Ring Detection Timeout m FXO Supplementary Services e __Hold Transfer Toward the Tel side e Blind Transfer to the Tel side e __Hold Transfer Toward the IP side SIP User s Manual 226 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 4 2 1 Automatic Dialing Automatic dialing is defined using the ini file parameter TargetOfChannelX where X is the channel number or the embedded Web server s Automatic Dialing screen refer to Section 5 5 9 2 on page 120 The SIP call flow diagram below illustrates Automatic Dialing Figure 8 3 Call Flow for Automatic Dialing SIP Client F1 INVITE Sent immediately if Caller ID detected otherwise sent after 2 rings or after 1 ring
443. r for FXS gateways is a special order option SIP User s Manual 334 Document LTRT 65408 SIP User s Manual 16 Selected Technical Specifications Table 16 1 MP 11x Functional Specifications continues on pages 333 to 335 Connectors amp Switches Rear Panel 8 Analog Lines MP 118 4 Analog Lines MP 114 2 Analog Lines MP 112 AC power supply socket Ethernet RS 232 Reset Button Physical Dimensions HxWxD Weight Environmenial Mounting Electrical Type Approvals Safety and EMC Management Configuration Management and Maintenance 8 RJ 11 connectors 4 RJ 11 connectors 2 RJ 11 connectors 100 240 0 3A max 10 100 Base TX RJ 45 Console PS 2 port Resets the MP 11x 42 x 172 x 220 mm 0 5 kg Approx Operational 5 to 40 C 41 to 104 F Storage 25 to 70 C 77 to 158 F Humidity 10 to 90 non condensing Rack mount Desktop Wall mount Note The rack mount is a special order option 100 240 VAC Nominal 50 60 Hz UL 60950 1 FCC part 15 Class B CE Mark EN 60950 1 EN 55022 EN 55024 EN61000 3 2 EN61000 3 3 EN55024 Gateway configuration using Web browser or ini files SNMP v2c SNMP v3 Syslog per RFC 3164 Local RS 232 terminal Web Management via HTTP or HTTPS Telnet Version 5 0 335 December 2006 Ta WH wt AudioCodes MediaPack 16 2 MP 124 Specifications Table 16 2 MP 124 Functional Specifications continues on pages 336 to 338 Chan
444. r telecommunication line cords Warning Units must be connected by service personnel to a socket outlet with a protective earthing connection Figure 3 4 RJ 45 Ethernet Connector Pinouts RJ 45 Connector and Pinout 12345678 eo not connected Figure 3 5 RJ 11 Phone Connector Pinouts RJ 11 Connector and Pinout 1234 Not connected Tip 1 2 3 Ring 4 Not connected Version 5 0 33 December 2006 7a Ta et AudioCodes MediaPack 3 1 5 1 Connecting the MP 11x RS 232 Port to Your PC Using a standard RS 232 straight cable not a cross over cable with DB 9 connectors connect the MP 11x RS 232 port using a DB 9 to PS 2 adaptor to either COM1 or COM2 RS 232 communication port on your PC The pinouts of the PS 2 connector is shown below in Figure 3 6 A PS 2 to DB 9 adaptor is not included with the MP 11x package For the PS 2 to DB 9 pinouts refer to Figure 3 7 below For information on establishing a serial communications link with the MP 11x refer to Section 10 2 on page 262 Figure 3 6 PS 2 Connector Pinouts PS 2 Female Connector and Pinout a 2 TD Transmit Data 1 Qs 3 GND Ground for Voltage ae 6 RD Receive Data Figure 3 7 PS 2 to DB 9 Adaptor Pinouts DB 9 Female PS 2 Male 2 2 3 4 6 5 4 3 3 1 5 2 Cabling the MP 11x FXS Lifeline The Lifeline provides a wired analog POTS phone connection to any PSTN or PBX FXS port when there is no power or when the
445. r Support Product Documentation Copyright 2006 AudioCodes Lid All rights reserved This document is subject to change without notice Date Published Dec 07 2006 Date Printed Dec 12 2006 When viewing this manual on CD Web site or on any other electronic copy all cross references are hyperlinked Click on the page or section numbers shown in blue to reach the individual cross referenced item directly To return back to the point from where you accessed the cross reference press the ALT and lt keys Trademarks AC logo Ardito AudioCoded AudioCodes AudioCodes logo IPmedia Mediant MediaPack MP MLQ NetCoder Stretto TrunkPack VoicePacketizer and VolPerfect are trademarks or registered trademarks of AudioCodes Limited All other products or trademarks are property of their respective owners 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
446. r is set to 100 and the Presentation is set to Restricted 1 Don t Play 0 Ringback tone isn t played to the IP side of the call default Play 1 Ringback tone is played to the IP side of the call after SIP 183 session progress response is sent applies only to FXS gateways in FXO gateways the Ringback tone isn t played Note 1 To enable the gateway to send a 183 response set EnableEarlyMedia to 1 Note 2 If EnableDigitDelivery 1 the gateway doesn t play a Ringback tone to IP and doesn t send a 183 response Don t Play 0 Ringback Tone isn t played Always Play 1 Ringback Tone is played to the Tel side of the call when 180 183 response is received Play According to 180 183 2 Ringback Tone is played to the Tel side of the call if no SDP is received in 180 183 responses If 180 183 with SDP message is received the gateway cuts through the voice channel and doesn t play Ringback tone default Play According to PI 3 N A SIP User s Manual 62 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter Enable GRUU EnableGRUU Use Tgrp Information UseSIPTgrp User Agent Information UserAgentDisplayInfo Subject SIPSubject Enable Reason Header EnableReasonHeader Description Determines whether or not the GRUU mechanism is used Valid options i
447. r it is available Note To use ProxyRedundancyMode enable Keep alive with Proxy option EnableProxyKeepAlive 1 or 2 Is Proxy Trusted This parameter isn t applicable and must always be set to Yes 1 IsTrustedProxy The parameter AssertedidMode should be used instead Enable Registration No 0 Gateway doesn t register to Proxy Registrar default IsRegisterNeeded Yes 1 Gateway registers to Proxy Registrar when the device is powered up and every RegistrationTime seconds Note The gateway sends a REGISTER request for each channel or for the entire gateway according to the AuthenticationMode parameter Registrar Name Registrar Domain Name RegistrarName If specified the name is used as Request URI in REGISTER messages If isn t specified default the Registrar IP address or Proxy name or Proxy IP address is used instead Registrar IP Address IP address and optionally port number of Registrar server RegistrarIP Enter the IP address in dotted format notation for example 201 10 8 1 lt 5080 gt Note 1 If not specified the REGISTER request is sent to the primary Proxy server refer to Proxy IP address parameter Note 2 When port number is specified DNS NAPTR SRV queries aren t performed even if DNSQueryType is set to 1 Registration Time Defines the time in seconds for which registration to a Proxy server is valid The RegistrationTime value is used in the header Expires
448. r s Manual 12 Security Table 12 3 SPD Table Configuration Parameters continues on pages 284 to 285 Parameter Name Description First to Fourth Proposal Determines the encryption type used in the quick mode negotiation for up to Encryption Type four proposals IPSecPolicyProposalEncrypt X stands for the proposal number 0 to 3 ion_X The valid encryption values are Not Defined default None 0 No encryption DES CBC 1 Triple DES CBC 2 AES 3 First to Fourth Proposal Determines the authentication protocol used in the quick mode negotiation for Authentication Type up to four proposals IPSecPolicyProposalAuthent X stands for the proposal number 0 to 3 ication_X The valid authentication values are Not Defined default HMAC SHA 1 96 2 HMAC MD5 96 4 If no IPSec methods are defined Encryption Authentication the default settings shown in Table 12 4 below are applied Table 12 4 Default IKE Second Phase Proposals Encryption Authentication Proposal 0 3DES SHA1 Proposal 1 3DES MD5 Proposal 2 DES SHA1 Proposal 3 DES MD5 gt To configure the SPD table using the ini file SPD table is configured using ini file tables described in Section 10 5 on page 267 Each line in the table refers to a different IP destination The Format line SPD_INDEX in the example below specifies the order in which the actual data lines are written The order of the parameters is irrelevant Parameters are not mandatory unle
449. r to the Startup process described in Figure 10 3 on page 266 E Ata configurable time of the day e g 18 00 This option is disabled by default m At fixed intervals e g every 60 minutes This option is disabled by default The following ini file example can be used to activate the Automatic Update mechanism Figure 10 2 Example of an ini File Activating the Automatic Update Mechanism DNS is required for specifying domain names in URLs DnsPriServerIP 10 1 1 11 Load an extra configuration ini file using HTTP IniFileURL http webserver corp com AudioCodes inifile ini Load Call Progress Tones file using HTTPS CptPileUrl https 10 31 2 17 use_tones dat Load Voice Prompts file using FTPS with user root and password wheel VPFileUrl ftps root wheel ftpserver corp com vp dat Update every day at 03 00 AM AutoUpdatePredefinedTime 03 00 Note The cmp file isn t updated since it is disabled by default AutoUpdateCmpFile Refer to the following notes E When HTTP or HTTPS are used the gateway contacts the Web server s and queries for the requested files The ini file is loaded only if it was modified since the last automatic update The cmp file is loaded only if its version is different from the version stored on the gateway s non volatile memory All other auxiliary files e g CPT are updated only once To update a previously loaded auxiliary file you must update the
450. r without seizing the line The line is seized only after the call is answered e After a ring signal is detected on a Disabled or Hotline FXO port the gateway seizes the line SIP User s Manual 120 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 28 Automatic Dialing ini File Parameter Parameter Name in ini File Parameter Format TargetOfChannelX TargetOfChannel lt Port gt lt Phone gt lt Mode gt For example TargetOfChannelO 1001 1 TargetOfChannel3 911 2 Note 1 The numbering of channels starts with 0 Note 2 Define this parameter for each gateway port you want to use for Automatic Dialing Note 3 This parameter can appear up to 8 times for 8 port gateways and up to 24 times for MP 124 gateways 5 5 9 3 Caller ID Use the Caller Display Information screen to send to IP Caller ID information when a call is made using the VoIP gateway relevant to both FXS and FXO The person receiving the call can use this information for caller identification The information on this table is sent in an INVITE message in the From header For information on Caller ID restriction according to destination source prefixes refer to Section 5 5 3 on page 91 Note If Caller ID name is detected on an FXO line EnableCallerlD 1 it is used instead of the Caller ID name defined in this table FXO gateways only To configure the Caller ID table take these 6 steps Open t
451. ral Media Settings according to Table 5 46 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 46 Media Settings General Media Settings Parameters Parameter DSP Version Template Number Max Echo Canceller Length MaxEchoCancellerLength Enable Continuity Tones Description N A Maximum Echo Canceller Length in msec 0 based on internal gateway settings default 4 32 msec 11 64 msec Note 1 The gateway must be reset after the value of MaxEchoCancellerLength is changed Note 2 It is unnecessary to configure the parameter EchoCancellerLength as it automatically acquires its value from the parameter MaxEchoCancellerLength N A Version 5 0 161 December 2006 Ta WH wt AudioCodes MediaPack 5 6 2 6 Media Settings ini File Parameters Table 5 47 describes the Media Settings parameters that can only be configured via the ini file Table 5 47 Media Settings ini File Parameters continues on pages 162 to 163 ini File Parameter Name Valid Range and Description RTPSIDCoeffNum Determines the number of spectral coefficients added to an SID packet being sent according to RFC 3389 Valid only if EnableStandardS IDPayloadT ype is set to 1 The valid values are 0 default 4 6 8 and 10 ECHybridLoss Sets the four wire to two wire worst case Hybrid loss the ratio betwee
452. rates this response in Call Waiting service When SIP gateway receives a 182 response it plays a special waiting Ringback tone to TEL side 183 Session Yes The SIP gateway generates this response if Early Media feature is Progress enabled and if the gateway plays a Ringback tone to IP B 5 2 2xx Response Successful Responses Table B 6 2xx SIP Responses 2xx Response Supported Comments 200 OK Yes 202 Accepted Yes SIP User s Manual 344 Document LTRT 65408 SIP User s Manual B SIP Compliance Tables B 5 3 3xx Response Redirection Responses 3xx Response 300 301 302 305 380 Multiple Choice Moved Permanently Moved Temporarily Use Proxy Alternate Service Supported Yes Yes Yes Yes Yes Table B 7 3xx SIP Responses Comments The gateway responds with an ACK and resends the request to first in the contact list new address The gateway responds with an ACK and resends the request to new address The SIP gateway generates this response when call forward is used to redirect the call to another destination If such response is received the calling gateway initiates an INVITE message to the new destination The gateway responds with an ACK and resends the request to new address B 5 4 4xx Response Client Failure Responses Table B 8 4xx SIP Responses continues on pages 345 to 346 4xx Response 400 401 402 403 404 405 406 407 4
453. rce varbind text for all the alarms under this component is Board lt n gt EthernetLink 0 where n is the slot number This trap relates to the Ethernet Link Module the 0 numbering doesn t apply to the physical Ethernet link Table F 10 acBoardEthernetLinkAlarm Alarm Trap Alarm acBoardEthernetLinkAlarm OID 1 3 6 1 4 1 5003 9 10 1 21 2 0 10 Default Severity Critical Event Type equipmentAlarm Probable Cause underlyingResourceUnavailable 56 Alarm Text Ethernet link alarm lt text gt Status Changes Condition Fault on single interface Alarm status major lt text gt value Redundant link is down Condition Fault on both interfaces Alarm status critical lt text gt value No Ethernet link Condition Both interfaces are operational Alarm status cleared Corrective Action Ensure that both Ethernet cables are plugged into the back of the system Inspect the system s Ethernet link lights to determine which interface is failing Reconnect the cable or fix the network problem Version 5 0 379 December 2006 7a VT e wt AudioCodes MediaPack F 1 4 Log Traps Notifications This section details traps that are not alarms These traps are sent with the severity varbind value of indeterminate These traps don t clear they don t appear in the alarm history or active tables One log trap that does send clear is acPerformanceMonitoringThresholdCrossing Table F 11 acKeepAlive Log Trap Trap acKeepAlive OID 1
454. rding to the parameter SessionExpiresMethod The default is O not activated Minimum Session Expires Defines the time in seconds that is used in the Min SE header This header MINSE defines the minimum time that the user agent supports for session refresh The valid range is 10 to 100000 The default value is 90 SIP User s Manual 58 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter Session Expires Method SessionExpiresMethod Asserted Identity Mode AssertedldMode Fax Signaling Method IsFaxUsed Detect Fax on Answer Tone DetFaxOnAnswerTone SIP Transport Type SIPTransportType Description Defines the SIP method used for session timer updates Re INVITE 0 Use Re INVITE messages for session timer updates default UPDATE 1 Use UPDATE messages Note 1 The gateway can receive session timer refreshes using both methods Note 2 The UPDATE message used for session timer purposes is not included in the SDP body Disable 0 None default Adding PAsserted Identity 1 Adding PPreferred Identity 2 The Asserted ID mode defines the header that is used in the generated INVITE request The header also depends on the calling Privacy allowed or restricted The P asserted or P preferred headers are used to present the originating party s Caller ID The Caller ID is composed of a Calli
455. reason for IP gt Tel calls is provided in Q 931 notation The release reason for Tel gt IP calls is provided in SIP 4xx 5xx and 6xx response codes For Tel gt IP calls an alternative IP address for IP gt Tel calls an alternative hunt group Refer to 5 5 5 2 on page 100 for information on defining an alternative IP address Refer to the 5 5 5 3 on page 102 for information on defining an alternative hunt group You can use this table for example E For Tel gt IP calls when there is no response to an INVITE message after INVITE retransmissions and the gateway then issues an internal 408 No Response implicit release reason E For IP gt Tel calls 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 DefaultReleaseCause in Table 5 8 Note The reasons for alternative routing option for Tel gt IP calls only applies when Proxy isn t used gt To configure the reasons for alternative routing take these 5 steps 1 Open the Reasons for Alternative Routing screen Protocol Management menu gt Routing Tables submenu gt Reasons for Alternative Routing option the Reasons for Alternative Routing screen is displayed Figure 5 21 Reasons for Alternative Routing Screen Reasons for Redundant Routing P to Tel Reasons Reason 1 Reason 2 Reason 3 Reason 4 el to IP Reasons Reason
456. receiving this message the GW issues a new request according to the scheme received on this response The gateway generates this response if the no answer timer expires On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call Version 5 0 345 December 2006 Ta fa AudioCodes MediaPack Table B 8 4xx SIP Responses continues on pages 345 to 346 4xx Response 409 411 413 414 415 420 480 481 482 483 484 485 486 487 488 Conflict Gone Length Required Request Entity Too Large Request URL Too Long Unsupported Media Bad Extension Temporarily Unavailable Call Leg Transaction Does Not Exist Loop Detected Too Many Hops Address Incomplete Ambiguous Busy Here Request Canceled Not Acceptable Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Comments The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway resp
457. releases a call when a second polarity reversal signal is detected Disable 0 Disable the current disconnect service default Enable 1 Enable the current disconnect service If the current disconnect service is enabled the FXO gateway releases a call when current disconnect signal is detected on its port while the FXS gateway generates a Current Disconnect Pulse after a call is released from IP The current disconnect duration is determined by the parameter CurrentDisconnectDuration The current disconnect threshold FXO only is determined by the parameter CurrentDisconnectDefaultThreshold The frequency at which the analog line voltage is sampled is determined by the parameter TimeToSampleAnalogLineVoltage No 0 Don t release the call Yes 1 Call is released if RTP packets are not received for a predefined timeout default Note 1 If enabled the timeout is set by the parameter BrokenConnectionEventTimeout in 100 msec resolution The default timeout is 10 seconds BrokenConnectionEventTimeout 100 Note 2 This feature is applicable only if RTP session is used without Silence Compression If Silence Compression is enabled the gateway doesn t detect that the RTP connection is broken Note 3 During a call if the source IP address from where the RTP packets were sent is changed without notifying the gateway the gateway filters these RTP packets To overcome this issue set Disco
458. represents all numbers from 5551200 to 5551300 2 3 4 xxx represents four digit numbers that start with 2 3 or 4 54324 represents any number that starts with 54324 54324xx represents a 7 digit number that starts with 54324 123 100 200 represents all numbers from 123100 to 123200 The VoIP gateway matches the rules starting at the top of the table For this reason enter more specific rules above more generic rules For example if you enter 551 in entry 1 and 55 in entry 2 the VoIP gateway applies rule 1 to numbers that starts with 551 and applies rule 2 to numbers that start with 550 552 553 554 555 556 557 558 and 559 However if you enter 55 in entry 1 and 551 in entry 2 the VoIP gateway applies rule 1 to all numbers that start with 55 including numbers that start with 551 Version 5 0 95 December 2006 Ta Ce AudioCodes MediaPack 5 5 4 Mapping NPI TON to Phone Context The Phone Context table is used to configure the mapping of NPI and TON to the Phone Context SIP parameter When a call is received from the ISDN the NPI and TON are compared against the table and the 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 You can also configure the Phone Context table using the ini file parameter Pho
459. required Version 5 0 377 December 2006 Ta fal AudioCodes MediaPack Table F 6 acBoardCallResourcesAlarm Alarm Trap Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status Note Condition Alarm status acBoardCallResourcesAlarm 1 3 6 1 4 1 5003 9 10 1 21 2 0 8 Major processingErrorAlarm softwareError 46 Call resources alarm Number of free channels exceeds the predefined RAI high threshold Major To enable this alarm the RAI mechanism must be activated EnableRAI 1 Number of free channels falls below the predefined RAI ow threshold Cleared Table F 7 acBoardControllerFailureAlarm Alarm Trap Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status Additional info Condition Alarm status Alarm OID Default Severity Event Type Probable Cause Alarm Text Status Changes Condition Alarm status Condition Alarm status acBoardControllerFailureAlarm 1 3 6 1 4 1 5003 9 10 1 21 2 0 9 Minor processingErrorAlarm softwareError 46 Controller failure alarm Proxy has not been found Major Proxy not found Use internal routing or Proxy lost looking for another Proxy Proxy is found The clear message includes the IP address of the located Proxy Cleared Table F 8 acBoardOverloadAlarm Alarm Trap acBoardOverloadAlarm 1 3 6 1 4 1 5003 9
460. revious 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 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 Relevant Parameters The following parameters described in Table 5 16 are used to configure the Alternative Routing mechanism E AltRoutingTel2IPEnable E AltRoutingTel2IPMode E IPConnQoSMaxAllowedPL E IPConnQoSMaxAllowedDelay Version 5 0 231 December 2006 7a VT wt AudioCodes MediaPack 8 8 8 9 Mapping PSTN Release Cause to SIP Response The MediaPack FXO gateway is used to interoperate 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 gt Tel calls The converse is also true For Tel IP calls the SIP 4xx or 5xx responses are mapped to tones played to the PSTN PBX When establishing an IP gt Tel call the following rules are applied If the remote party PSTN PBX is busy and the FXO gateway detects a Busy tone it sends 486 busy to IP If it detects a Reorder tone it sends 404 not found no route to destination to IP In both cases the call is releas
461. roup1 and SecurityModel usm Delete the row in the usmUserTable for v3admin1 14 8 3 Trusted Managers By default the agent accepts get and set requests from any IP address as long as the correct community string is used in the request Security can be enhanced via the use of Trusted Managers A Trusted Manager is an IP address from which the SNMP Agent accepts and processes get and set requests An EM can be used to configure up to five Trusted Managers Note If Trusted Managers are defined all community strings work from all Trusted Managers That is there is no way to associate a community string with particular trusted managers The concept of trusted managers is considered to be a weak form of security and is therefore not a required part of SNMPv3 security which uses authentication and privacy However the board s SNMP agent applies the trusted manager concept as follows m There is no way to configure trusted managers for only a SNMPv3 user An SNMPv2c community string must be defined E If specific IPs are configured as trusted managers via the community table then only SNMPv3 users on those trusted managers are given access to the agent s MIB objects Version 5 0 317 December 2006 7a Ta wt AudioCodes MediaPack 14 8 3 1 Configuration of Trusted Managers via ini File To set the Trusted Mangers table from start up write the following in the ini file SNMPTRUSTEDMGR_X D D D
462. roximately 60 seconds to complete e Sets port 1 offhook listens to the dial tone e Sets port 1 and port 2 offhook dials the number of port 2 listens to the busy tone e Sets port 1 offhook dials the number of port 2 listens to the Ringback tone e Sets port 1 offhook dials an invalid number listens to the reorder tone Version 5 0 369 December 2006 gt WH wi AudioCodes MediaPack 2 The wizard then analyzes the recorded Call Progress Tones and displays a message specifying the tones that were detected by the gateway and analyzed by the wizard correctly At the end of a successful detection operation the detected Call Progress Tones are displayed in the Tones Analyzed pane refer to Figure E 8 Figure E 8 Recording Screen after Automatic Detection E AudioCodes Call Progress Tones Wizard 2 x Automatic Manual m Automatic tone detection and analysis Start Automatic Configuration Automatic analysis complete Tones analyzed 4 of 4 The gateway correctly detected 4 of 4 tones HEBER RBD ORD SDRH DORE SRR EORDERRE one Tones analyzed Dial Tone auto 350 Busy Tone aut 480 Ringing Tone 450 Reorder Tone 480 AudioCodes Cancel 3 All four Call Progress Tones are saved as standard A law PCM at 8000 bits per sample in the same directory as the CPTWizard exe file is located with the following names e cpt_recorded_dialtone pcm e cpt_recorded_bu
463. rridden by configuring the Presentation parameter in the Source Number Manipulation table 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 4 Repeat steps 2 and 3 for each VoIP gateway port 5 Click the Submit button to save your changes 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 29 Caller ID ini File Parameter Parameter Name in ini File Parameter Format CallerDisplayInfoX CallerDisplaylnfo lt Port gt lt Caller ID string gt lt Restriction gt 0 Not restricted default 1 Restricted For example CallerDisplaylnfoO Susan C 0 CallerDisplaylnfo2 Mark M 1 Note 1 The numbering of channels starts with 0 Note 2 This parameter can appear up to eight times for 8 port gateways and up to 24 times for MP 124 5 5 9 4 Generate Caller ID to Tel The Generate Caller ID to Tel screen for FXS and Caller ID Permissions screen for FXO are used to enable or disable per port the Caller ID generation for FXS gateways and detection for FXO gateways If a port isn t configured its Caller ID generation detection are determined according to the global parameter EnableCallerID described in Table 5 9 SIP User s Manual 122 Document LTRT 65408 SIP User s Manual 5 Web
464. rs can assign different Profiles behavior on a per call basis using the Tel to IP and IP to Hunt Group Routing tables or associate different Profiles to the gateway s endpoint s The Profiles contain parameters such as Coders T 38 Relay Voice and DTMF Gains Silence Suppression Echo Canceler RTP DiffServ Current Disconnect and more The Profiles feature allows users to tune these parameters or turn them on or off per source or destination routing and or the specific gateway or its ports For example specific ports can be designated to have a profile which always uses G 711 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 are applied The default values of the parameters in the Tel and IP Profiles are identical to the Web ini file parameter values If a value of a parameter is changed in the Web ini file it is automatically updated in the Profiles correspondingly After any parameter in the Profile is modified by the user modifications to parameters in the Web ini file no longer impact that Profile 5 5 6 1 Coder Group Settings Use the Coder Group Settings screen to define up to four different coder groups These coder groups
465. rsion Utility to m Create a loadable Call Progress Tones file refer to Section E 1 1 on page 364 m Encode decode an ini file refer to Section E 1 2 on page 365 E Create a loadable Prerecorded Tones file refer to Section E 1 3 on page 366 Figure E 1 TrunkPack Downloadable Conversion Utility Opening Screen te TrunkPack Downloadable Conversion Utility R2 5 2 Process Call Progress Tones file s Process Voice Prompts file s Process CAS Tables Process VXML file s Process Prerecorded Tones file s Process Encoded Decoded ini file s Process Coder Description file s Version 5 0 363 December 2006 7a VT r wt AudioCodes MediaPack E 1 1 Converting a CPT ini File to a Binary dat File For detailed information on creating a CPT ini file refer to Section 15 1 on page 325 gt To convert a CPT ini file to a binary dat file take these 10 steps 1 Execute the TrunkPack Downloadable Conversion Utility DConvert exe supplied with the software package the utility s main screen opens shown in Figure E 1 2 Click the Process Call Progress Tones File s button the Call Progress Tones screen shown in Figure E 2 opens Figure E 2 Call Progress Tones Conversion Screen Call Progress Tones i x rm Call Progress Tones File Chu Select File me Using File c Media Gateway usa_tones ini Output File c media gateway usa_tones dat User Data Vendor Company Version h
466. rtificate To solve this add the IP address and host name ACL_nnnnnn where nnnnnn is the serial number of the MediaPack to your hosts file located at etc hosts on UNIX or C Windows System32 Drivers ETC hosts on Windows then use the host name in the URL e g https ACL_280152 The figure below is an example of a host file Figure 12 8 Example of a Host File This is a sample HOSTS file used by Microsoft TCP IP for Windows Location C WINDOWS SYSTEM32 DRIVERS ETC hosts LAT 10 Wig al localhost 10 31 4 47 ACL_280152 12 2 3 Secured Telnet To enable the embedded Telnet server on the MediaPack set the parameter TelnetServerEnable described in Table 5 37 on page 142 to 1 standard mode or 2 SSL mode no information is transmitted in the clear when SSL mode is used If the Telnet server is set to SSL mode a special Telnet client is required on your PC to connect to the Telnet interface over a secured connection examples include C Kermit for UNIX Kermit 95 for Windows and AudioCodes acSSLTelnet utility for Windows that requires prior installation of the free OpenSSL toolkit Contact AudioCodes to obtain the acSSLtTelnet utility Version 5 0 289 December 2006 Ta A wi AudioCodes MediaPack 12 2 4 Server Certificate Replacement The MediaPack is supplied with a working SSL configuration consisting of a unique self signed server certificate If an organizational Public Key Infrastructure PKI is u
467. s User agent redirect or registrar servers typically use 401 Unauthorized response to challenge authentication containing a WWW Authenticate header and expect the re INVITE to contain an Authorization header The following example describes the Digest Authentication procedure including computation of user agent credentials The REGISTER request is sent to Registrar Proxy server for registration as follows REGISTER sip l0 2 2 222 FESTE viag SIDHA A U 10 1 1 200 mamme lt eaas l22C TOR yin 2N eml ea Tor lt sipy 127e 10 1 1 2002 Call ID 634293194 10 1 1 200 User Agent Audiocodes Sip Gateway MP 118 FXS v 4 20 299 410 CSeq 1 REGISTER Contact asip ZZAN EEAO Expires 3600 SIP User s Manual 240 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities On receiving this request the Registrar Proxy returns 401 Unauthorized response SIP 2 0 401 Unauthorized Via Sue 2 OUDER 10 2 1 200 ETOM KEajos 2A A 2 222 SpiceioHieil 7OAo Woe lt SayjoslePVCiO 2oPsBZe gt Cern SIAAISLGACELO i i 206 Cseq 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 11432d6bce58ddf02e3b5elc77c010d2 stale FALSE algorithm MD5 According to the sub header present in the WWW Authenticate header the correct REGISTER request is formed Since the algorithm used is MD5 take The username is
468. s The startup process illustrated in Figure 10 3 on page 266 begins when the gateway is reset physically or from the Web SNMP and ends when the operational software is running In the startup process the network parameters software and configuration files are obtained After the gateway powers up or after it is physically reset it broadcasts a BootRequest message to the network If it receives a reply from a BootP server it changes its network parameters IP address subnet mask and default gateway address to the values provided If there is no reply from a BootP server and if DHCP is enabled DHCPEnable 1 the gateway initiates a standard DHCP procedure to configure its network parameters After changing the network parameters the gateway attempts to load the cmp and various configuration files from the TFTP server s IP address received from the BootP DHCP servers If a TFTP servers IP address isn t received the gateway attempts to load the software cmp file and or configuration files from a preconfigured TFTP server refer to Section 10 3 on page 263 Thus the gateway can obtain its network parameters from BootP or DHCP servers and its software and configuration files from a different TFTP server preconfigured in ini file If BootP DHCP servers are not found or when the gateway is reset from the Web SNMP it retains its network parameters and attempts to load the software cmp file and or configuration files fro
469. s to finish e Review the new subnet mask and press 1 to save it To change the default gateway IP address press 3 followed by the key e The gateway s current default gateway address is played To change press e Dial the new default gateway address e g 192 168 0 1 and press to finish e Review the new default gateway address and press 1 to save it Hang up the handset Access the gateway s Embedded Web Server with the new IP address you assigned refer to Section 5 3 on page 51 Complete the gateway s configuration and save it to non volatile memory refer to Section 5 10 2 on page 205 The following configuration parameters can be queried or modified via the voice menu Item Number at Menu Prompt 1 2 3 4 7 11 Table 4 2 Configuration Parameters Available via the Voice Menu continues on pages 43 to 44 Description IP address Subnet mask Default gateway IP address Primary DNS server IP address DHCP enable disable MGCP call agent IP address N A Version 5 0 43 December 2006 Ta WH wt AudioCodes MediaPack Table 4 2 Configuration Parameters Available via the Voice Menu Item Number at Menu Prompt 12 99 continues on pages 43 to 44 Description MGCP call agent port number N A Voice menu password initially 12345 Note The voice menu password can also be changed using the parameter VoiceMenuPassword refer to Table 5 50 on page 174 4 2 4 Assign
470. s Packets sent in this phase have two octets of PPP header followed by LCP message with variable length Various parameters and options are negotiable at this phase including MRU maximum receive unit Authentication Protocol and others Once the link is established each side sends a configure ack message to the other side the authentication phase may begin The authentication phase is not mandatory However it is negotiated in the link configuration phase A host may ask other hosts for authentication using Password Authentication Protocol PAP or Challenge Handshake Authentication Protocol CHAP The PAP sends the username and password to the remote host unencrypted The CHAP is a more sophisticated method of authentication The two hosts share a secret The authenticator sends a challenge to the host requesting authentication The host performs a calculation one way hash using the challenge received from the authenticator and the shared secret and sends the result to the authenticator The authenticator verifies the host if the result of the calculation is correct otherwise it is rejected The last configuration phase immediately after the authentication phase or after the Link Configuration is the Network Control Protocol There is a family of control protocols for establishing and configuring different network layer protocols for example IPCP PPP Internet Protocol Control Protocol IPv6 CP PP
471. s Manual 272 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities gt To replace the default logo with your own corporate image via the ini file take these 2 steps 1 Place your corporate logo image file in the same folder as where the device s ini file is located i e the same location defined in the BootP TFTP configuration utility For detailed information on the BootP TFTP refer to Appendix C on page 349 2 Add modify the two ini file parameters in Table 10 3 according to the procedure described in Section 6 2 on page 209 Note Loading the device s ini file via the Configuration File screen in the Web Interface doesn t load the corporate logo image files as well Table 10 3 Customizable Logo ini File Parameters Parameter Description LogoFileName The name of the image file containing your corporate logo Use a gif jpg or jpeg image file The default is AudioCodes logo file Note The length of the name of the image file is limited to 47 characters LogoWidth Width in pixels of the logo image Note The optimal setting depends on the resolution settings The default value is 339 which is the width of AudioCodes displayed logo 10 6 1 2 Replacing the Main Corporate Logo with a Text String The main corporate logo can be replaced with a text string m To replace AudioCodes default logo with a text string via the Web Interface modify the two ini file parameters in
472. s Parameters continues on pages 84 to 86 Parameter Time Between Call Waiting Indications TimeBetweenWaitingIndi cations Time before Waiting Indication TimeBeforeWaitinglIndica tion Waiting Beep Duration WaitingBeepDuration Enable Caller ID EnableCallerID Caller ID Type CallerIDType Hook Flash Code HookFlashCode MWI Parameters Enable MWI EnableMWI MWI Analog Lamp MWIAnalogLamp MWI Display MWIDisplay Description Difference in seconds between call waiting indications FXS only for call waiting The default value is 10 seconds Defines the interval in seconds before a call waiting indication is played to the port that is currently in a call FXS only The valid range is 0 to 100 The default time is 0 seconds Duration in msec of waiting indications that are played to the receiving side of the call FXS only for Call Waiting The default value is 300 No 0 Disable the Caller ID service default Yes 1 Enable the Caller ID service If the Caller ID service is enabled then for FXS gateways calling number and Display text are sent to gateway port For FXO gateways the Caller ID signal is detected and is sent to IP in SIP INVITE message as Display element For information on the Caller ID table refer to Section 5 5 9 3 on page 121 To disable enable caller ID generation per port refer to Section 5 5 9 4 on page 122 Defines one of the followi
473. s Tones levels and frequencies that the VoIP gateway uses The default CPT file is U S A Prerecorded Tones The dat PRT file enhances the gateway s capabilities of playing a wide range of telephone exchange tones that cannot be defined in the Call Progress Tones file User Information The User Information file maps PBX extensions to IP numbers This file can be used to 7 8 represent PBX extensions as IP phones in the global IP world To load an auxiliary file to the gateway take these 8 steps Open the Auxiliary Files screen Software Upgrade menu gt Load Auxiliary Files the Auxiliary Files screen is displayed Click the Browse button that is in the field for the type of file you want to load Navigate to the folder that contains the file you want to load Select the file and click the Open button the name and path of the file appear in the field beside the Browse button Click the Send File button that is next to the field that contains the name of the file you want to load An exclamation mark in the screen section indicates that the file s loading doesn t take effect on the fly e g CPT file Repeat steps 2 to 5 for each file you want to load Saving an auxiliary file to flash memory may disrupt traffic on the MediaPack To avoid this disable all traffic on the device before saving to flash memory by performing a graceful lock refer to Section 5 10 1 on page 204 A MediaPack reset is
474. s currently accessing files on that remote NFS file system Version 5 0 143 December 2006 Ta WH wt AudioCodes MediaPack To delete a remote NFS file system take these 3 steps gt 1 Select the Edit radio button for the row to be deleted 2 Click the Delete Line button the row is deleted 3 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 gt To modify an existing remote NFS file system take these 4 steps 1 Select the Edit radio button for the row to be modified 2 Change the values on the selected row according to your requirements 3 Click the Apply New Settings button the remote NFS file system is mounted using the new settings Check the Syslog server for the NFS mount was successful message 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 38 Network Settings NFS Settings Parameters Parameter Description Line Number The row index of the remote file system NFSServers_Index The valid range is 0 to 4 Host IP The domain name or IP address of the NFS server If a domain name NFSServers_HostOrIP is provided a DNS server must be configured Path to the root of the remote file system In the format path For example audio Root Path NFSServers_RootPath The combination of Host IP and Root Path must be unique for each row in the table For example there m
475. s displayed Figure 5 33 FXO Settings Screen FXO Settings Dialing Mode One Stage Waiting for Dial Tone Time to Wait before Dialing msec Ring Detection Timeout sec Reorder Tone Duration sec Answer Supervision Rings before Detecting Caller ID Send Metering Message to IP Disconnect on Busy Tone Disconnect On Dial Tone Guard Time Between Calls No 1000 8 D No 1 No Yes Disable 1 2 Configure the FXO parameters according to Table 5 33 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 33 FXO Parameters continues on pages 127 to 130 Parameter Dialing Mode IsTwoStageDial Description One Stage 0 One stage dialing Two Stage 1 Two stage dialing default Used for IP gt FXO gateways calls If two stage dialing is enabled then the FXO gateway seizes one of the PSTN PBX lines without performing any dial the remote user is connected over IP to PSTN PBX and all further signaling dialing and Call Progress Tones is performed directly with the PBX without the gateway s intervention If one stage dialing is enabled then the FXO gateway seizes one of the available lines according to Channel Select Mode parameter and dials the destination phone number received in INVI
476. s facility to set up the gateway configuration parameters Users also have the option to remotely reset the gateway and to permanently apply the new set of parameters 5 1 Computer Requirements To use the Embedded Web Server the following is required A computer capable of running your Web browser m A network connection to the VoIP gateway m One of the following compatible Web browsers e Microsoft Internet Explorer version 6 0 and higher e Netscape Navigator version 7 2 and higher Note The browser must be Java script enabled If java script is disabled access to the Embedded Web Server is denied 5 2 Protection and Security Mechanisms Access to the Embedded Web Server is controlled by the following protection and security mechanisms m User accounts refer to Section 5 2 1 below Read only mode refer to Section 5 2 2 on page 51 Disabling access refer to Section 5 2 3 on page 51 Secured HTTP connection HTTPS refer to Section 12 2 2 on page 288 Limiting access to a predefined list of IP addresses refer to Section 5 6 5 2 on page 170 m Managed access using a RADIUS server refer to Section 12 3 on page 293 5 2 1 User Accounts To prevent unauthorized access to the Embedded Web Server two user accounts are available a primary and secondary Each account is composed of three attributes username password and access level The username and password enable access to the Embedded We
477. sable NAT Settings NAT IP Address 0 0 0 0 Differential Services Network QoS 48 Media Premium QoS Control Premium QoS Gold QoS Bronze QoS 2 Configure the IP Settings according to Table 5 36 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 36 Network Settings IP Settings Parameters continues on pages 138 to 140 Parameter Description IP Networking Mode Enables disables the Multiple IPs mechanism EnableMultiplelPs Single IP Network 0 default Multiple IP Network 1 For detailed information on Multiple IPs refer to Section 9 8 on page 253 IP Address IP address of the gateway Enter the IP address in dotted format notation for example 10 8 201 1 Note 1 A warning message is displayed after pressing the button Submit if the entered value is incorrect Note 2 After changing the IP address and pressing the button Submit a prompt appears indicating that for the change to take effect the gateway is to be reset SIP User s Manual 138 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 36 Network Settings IP Settings Parameters continues on pages 138 to 140 Parameter Description Subnet Mask Subnet mask of the gateway Enter the subnet mask in dotted format notation for example 255 255 0 0 Note 1 A warning message
478. se Numeric Stop Acc cause P Start Acc 26 h323 gw id 33 Name of the gateway String SIPIDString Stop Acc di Called String 8004567145 Start Acc Station Id Destination phone number String 2427456425 Stop Acc Calling Start Acc 31 Station Id Calling Party Number ANI String 5135672127 Stop Acc Account Request Type start f E or stop 40 oe Note start isn t supported Numeric ao a ae as yp on the Calling Card P P application Acct Delay No of seconds tried in Start Acc Ai Time sending a particular record Numeric 9 Stop Acc Acct Input Number of octets received a Octets for that call duration Numeric Stop Acc Acct Output Number of octets sent for s ag Octets that call duration Numeric Stop Ace Acct Session A unique accounting Start Acc i Id identifier match start amp stop String 340a Stop Acc 46 Acct Session For how many seconds the Numene Stop Acc Time user received the service Acct Input Number of packets received ae Packets during the call Numgiig Stop Acc Acct Output Number of packets sent Packets during the call Numeric Stop Acc 3 MediaPack physical port 0 61 NAS Por type on which the call is String Asynchrono Sia Ae Type Stop Acc active us Response Attributes The reason for failing 26 No a reium 103 authentication 0 ok other Numeric O Request Stop Acc code accepted number failed A unique accounting 44 cea identifier match start amp String Stop Acc stop S
479. sec 1 the maximum number of simultaneous channels supported is 120 FaxModemBypassDJBufMinDe Determines the Jitter Buffer delay during fax and modem bypass session lay The valid range is 0 to 150 msec The default is 40 EchoCancellerAggressiveNLP Enables or disables the Aggressive NLP in the first 0 5 second of the call 0 Disabled default 1 Enabled SIP User s Manual 162 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 47 Media Settings ini File Parameters continues on pages 162 to 163 ini File Parameter Name NSEMode NSEPayloadType IsCiscoSCEMode BellcoreCallerIDTypeOneSubS tandard ETS CallerIDTypeOneSubStan dard ETSIVMWITypeOneStandard BellcoreVMWITypeOneStandar d Valid Range and Description Cisco compatible fax and modem bypass mode 0 NSE disabled default 1 NSE enabled Note 1 This feature can be used only if VxxModemTransportType 2 Bypass Note 2 If NSE mode is enabled the SDP contains the following line a rtpmap 100 X NSE 8000 Note 3 To use this feature The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw Set the Modem transport type to Bypass mode VxxModemTransportType 2 for all modems a Configure the gateway parameter NSEPayloadType 100 In NSE bypass mode the gateway starts using G 711 A Law default or G 711u Law according to the parameter FaxMo
480. seconds the VoIP gateway waits before forwarding the call to the phone number specified Parameter Format FwdlInfo_ lt Port 0 to 23 gt lt Forward Type gt lt Forwarded SIP User Identification gt lt Timeout in seconds for No Reply gt For example FwdlInfo_0 1 1001 Fwdlinfo_1 1 2003 10 5 1 1 Fwdlnfo_2 3 2005 30 Note 1 The numbering of gateway ports starts with 0 Note 2 This parameter can appear up to 24 times for MP 124 Version 5 0 125 December 2006 Ta A wi AudioCodes MediaPack 5 5 10 Configuring RADIUS Accounting Parameters The RADIUS Parameters screen is used for configuring the Remote Authentication Dial In User Service RADIUS accounting parameters gt To configure the FXO parameters take these 4 steps 1 Open the RADIUS Parameters screen Protocol Management menu gt RADIUS Parameters the RADIUS Parameters screen is displayed RADIUS Enable RADIUS Enable RADIUS Accounting Server IP Address 0 0 0 0 RADIUS Accounting Port 1646 RADIUS Accounting Type At Connect and Release AAA Indications Figure 5 32 RADIUS Parameters Screen Parameters None 2 Configure the RADIUS accounting parameters according to Table 5 33 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Parameter Enable R
481. sed you may wish to replace this certificate with one provided by your security administrator gt To replace the MediaPack self signed certificate take these 9 steps 1 Your network administrator should allocate a unique DNS name for the MediaPack e g dns_name corp customer com This name is used to access the device and should therefore be listed in the server certificate 2 Open the Certificates screen Advanced Configuration menu gt Security Settings submenu gt Certificates option the Certificates screen is displayed Figure 12 9 Figure 12 9 Certificate Signing Request Screen Certificate Signing Request Subject Name Copy the certificate signing request and send it to your Certification Authority for signing Certificate Files Send Server Certificate file from your computer to the device e Browse _ Send File Send Trusted Root Certificate Store file from your computer to the device e r i Browse _ Send file Send Private Key file from your computer to the device n Browse _ Send file Note Replacement of the private key is not recommended and should be done over a physically secure network link SIP User s Manual 290 Document LTRT 65408 SIP User s Manual 12 Security 3 In the Subject Name field enter the DNS name and click Generate CSR A textual certificate signing request that contains the SSL device identifier is disp
482. shortest path to a certain destination and signal the remote host the existence of the better route Using multiple router support the media gateway 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 SIP User s Manual 252 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities 9 7 Simple Network Time Protocol Support Simple Network Time Protocol SNTP client functionality generates 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 is able to synchronize 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 actions 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 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 should be every 24 hours based on when the system was restarted The NTP server i
483. sinnhiceabensimanionbaeaittnnanaviomieaemen ene SIP User s Manual 10 Document LTRT 65408 SIP User s Manual Contents List of Figures asad T T Hi MediaPack VoIP Ap i Version 5 0 _ 41 December 2006 MediaPack SIP User s Manual 12 Document LTRT 65408 SIP User s Manual Contents ett jure 1 5 1 C jure 15 2 jure 1 jure 1 jure 1 u u u ju ju u ur ur u Version 5 0 13 December 2006 7a et AudioCodes MediaPack List of Tables SIP User s Manual 14 E Document LTRT 65408 SIP User s Manual Contents 6 2xx SR pee 1S 7 Bx SIP Res Version 5 0 15 December 2006 ra wii AudioCodes MediaPack Table D 1 Pac Table Ta Tab Table Tabl Tab Table Ta ble SIP User s Manual 16 Document LTRT 65408 SIP User s Manual Notices 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 Updates to this document and other documents can be viewed by registered Technical Support customers at www audiocodes com unde
484. sion 5 0 287 December 2006 a WH wt AudioCodes MediaPack 12 2 12 2 1 12 2 2 SSL TLS SSL also known as TLS is the method used to secure the MediaPack SIP Signaling connections Embedded Web Server and Telnet server The SSL protocol provides confidentiality integrity and authenticity between two communicating applications over TCP IP Specifications for the SSL TLS implementation E Supports transports SSL 2 0 SSL 3 0 TLS 1 0 E Supports ciphers DES RC4 compatible m Authentication X 509 certificates CRLs are not supported SIP Over TLS SIPS The MediaPack uses TLS over TCP to encrypt SIP transport and optionally to authenticate it To enable TLS on the MediaPack set the selected transport type to TLS SIPTransportType 2 In this mode the gateway initiates a TLS connection only for the next network hop To enable TLS all the way to the destination over multiple hops set EnableSIPS to 1 When a TLS connection with the gateway is initiated the gateway also responds using TLS regardless of the configured SIP transport type in this case the parameter EnableSIPS is also ignored TLS and SIPS use the Certificate Exchange process described in Sections 12 2 4 and 12 2 5 To change the port number used for SIPS transport by default 5061 use the parameter TLSLocalSIPPort When SIPS is used it is sometimes required to use two way authentication When acting as the TLS server in a specific connection it
485. slog Can be disabled DisableSNMP Version 5 0 299 December 2006 Ta WH wt AudioCodes MediaPack 12 7 12 8 Recommended Practices To improve network security the following guidelines are recommended when configuring the MediaPack Set the password of the primary web user account refer to 5 6 5 1 on page 168 toa unique hard to hack string Do not use the same password for several devices as a single compromise may lead to others Keep this password safe at all times and change it frequently If possible use a RADIUS server for authentication RADIUS allows you to set different passwords for different users of the MediaPack with centralized management of the password database Both Web and Telnet interfaces support RADIUS authentication refer to Section 12 3 on page 293 If the number of users that access the Web and Telnet interfaces is limited you can use the Web and Telnet Access List to define up to ten IP addresses that are permitted to access these interfaces Access from an undefined IP address is denied refer to Section 5 6 5 2 on page 170 Use IPSec to secure traffic to all management and control hosts Since IPSec encrypts all traffic hackers cannot capture sensitive data transmitted on the network and malicious intrusions are severely limited Use HTTPS when accessing the Web interface Set HTTPSOnly to 1 to allow only HTTPS traffic and block port 80 If you don t need the Web int
486. special notes pertaining to MIBs A detailed explanation of each parameter can be viewed in an SNMP browser in the MIB Description field Not all groups in the MIB are functional Refer to version release notes Certain parameters are non functional Their MIB status is marked obsolete When a parameter is set to a new value via SNMP the change may affect device functionality immediately or may require that the device be soft reset for the change to take effect This depends on the parameter type The current updated device configuration parameters are programmed into the device provided that the user does not load an ini file to the device after reset Loading an ini file after reset overrides the updated parameters Additional MIBs are to be supported in future releases Version 5 0 311 December 2006 Ta laS AudioCodes MediaPack 14 7 Traps Note As of this version all traps are sent from the SNMP port default 161 This is part of the NAT traversal solution Full proprietary trap definitions and trap Varbinds are found in the acBoard and acAlarm MIBs The following proprietary traps are supported For detailed information on these traps refer to Appendix F on page 375 Table 14 1 Proprietary Traps Description Trap acBoardFatalError acBoardConfigurationError acBoardTemperatureAlarm acBoardEvResettingBoard acBoardEvBoardStarted acgwAdminStateChange acOperationalStateChange acB
487. spelnia podstawowe wymagania i odpowiada warunkom zawartym w dyrektywie 89 336 EEC 73 23 EEC 1999 5 ES AudioCodes Ltd declara que este MP 11x FXS amp FXO Series amp MP 124 esta conforme com os requisitos essenciais e outras disposi es da Directiva 89 336 EEC 73 23 EEC 1999 5 ES AudioCodes Ltd t mto vyhlasuje e MP 11x FXS amp FXO Series amp MP 124 sp a z kladn po iadavky a v etky pr slu n ustanovenia Smernice 89 336 EEC 73 23 EEC 1999 5 ES iuo AudioCodes Ltd deklaruoja kad is MP 11x FXS amp FXO Series amp MP 124 atitinka esminius reikalavimus ir kitas 89 336 EEC 73 23 EEC 1999 5 ES Direktyvos nuostatas Por medio de la presente AudioCodes Ltd declara que el MP 11x FXS amp FXO Series amp MP 124 cumple con los requisitos esenciales y cualesquiera otras disposiciones aplicables o exigibles de la Directiva 89 336 EEC 73 23 EEC 1999 5 ES H rmed intygar AudioCodes Ltd att denna MP 11x FXS amp FXO Series amp MP 124 st r I verensst mmelse med de v sentliga egenskapskrav och vriga relevanta best mmelser som framg r av direktiv 89 336 EEC 73 23 EEC 1999 5 ES 1 Installation and service of this gateway must only be performed by authorized qualified service personnel 2 To avoid risk of fire use 26 AWG or higher wiring to connect the FXS or FXO telecom ports 3 The equipment must be connected by service personnel to a socket outlet with a protective earthing connecti
488. sponse User initiated BootP respond for remote provisioning over WAN Filtered display of BootP requests Location of other BootP utilities that contain the same MAC entity Common log window for both BootP and TFTP sessions Works with Windows 98 Windows NT Windows 2000 and Windows XP C 4 Specifications BootP standards RFC 951 and RFC 1542 TFTP standards RFC 1350 and RFC 906 Operating System Windows 98 Windows NT Windows 2000 and Windows XP Max number of MAC entries 200 C 5 Installation oh To install the BootP TFTP on your computer take these 2 steps Locate the BootP folder on the VoIP gateway supplied CD ROM and open the file Setup exe Follow the prompts from the installation wizard to complete the installation To open the BootP TFTP take these 2 steps From the Start menu on your computer navigate to Programs and then click BootP The first time that you run the BootP TFTP the program prompts you to set the user preferences Refer to the Section C 10 on page 353 for information on setting the preferences C 6 Loading the cmp File Booting the Device Once the application is running and the preferences were set refer to Section C 10 for each unit that is to be supported enter parameters into the tool to set up the network configuration information and initialization file names Each unit is identified by a MAC address For information on how to configure add delete and edit un
489. ss level the screen cannot be viewed m If the access level of the user is equal to or greater than the screen s read access level but less than the write access level the screen is read only m If the access level of the user is equal to or greater than the screen s write access level the screen can be modified The default attributes for the two accounts are shown in Table 5 2 below Table 5 2 Default Attributes for the Accounts Account Attribute Username Password Access Level Case Sensitive Case Sensitive Primary Account Admin Admin Security Administrator Secondary Account User User User Monitor The access level of the primary account cannot be changed all other account attributes can be modified The first time a browser request is made users are requested to provide their account s username and password to obtain access If the Embedded Web Server is left idle for more than five minutes the session expires and the user is required to re enter username and password Tip To access the Embedded Web Server with a different account click the Log Off button and then re access with a new username and password For details on changing the account attributes refer to Section 5 6 5 1 on page 168 Note that the password and username can be a maximum of 19 case sensitive characters To reset the username and password of both accounts to their defaults set the ini file parameter ResetWebPassword to 1
490. ss stated otherwise To support more than one Encryption Authentication proposals for each proposal specify the relevant parameters in the Format line Note that the proposal list must be contiguous Figure 12 4 Example of an SPD Table IPSEC_SPD_TABLE Format SPD_INDEX IPSecPolicyRemoteIPAddress IpsecPolicySrcPort IPSecPolicyDStPort IPSecPolicyProtocol IPSecPolicyLifelInSec IPSecPolicyProposalEncryption_0 IPSecPolicyProposalAuthentication_0 IPSecPolicyProposalEncryption_1 IPSecPolicyProposalAuthentication_l IPSecPolicyKeyExchangeMethodIndex IPSecPolicyLocallIPAddressType IPSMC_SED_ITAELT 19 11 2 21 0 O 17 S00 1 2 IPSEC_SPD_TABLE Version 5 0 285 December 2006 Ta A wl AudioCodes MediaPack In the SPD example all packets designated to IP address 10 11 2 21 that originates from the OAM interface regardless to their destination and source ports and whose protocol is UDP are encrypted the SPD also defines an SA lifetime of 900 seconds and two security proposals DES SHA1 and 3DES SHA1 gt To configure the SPD table using the Embedded Web Server take these 6 steps Access the Embedded Web Server refer to Section 5 3 on page 51 2 Open the IPSec Table screen Advanced Configuration menu gt Security Settings gt IPSec Table option the IPSec Table screen is displayed aoe Figure 12 5 IPSec Table Screen IPSec Table Policy Index fo State Does not e
491. sserted and P Preferred headers 0 sip default 1 tel Defines the time in seconds the gateway 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 Enable Remote Party ID RPI headers for calling and called numbers for Tel gt IP calls Disable 0 default Enable 1 RPI headers are generated in SIP INVITE messages for both called and calling numbers SIP User s Manual 60 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 3 Protocol Definition General Parameters continues on pages 58 to 64 Parameter Enable History Info Header EnableHistoryInfo Use Source Number as Display Name UseSourceNumberAsDis playName Description Enables usage of the History Info header Valid options include 0 Disable default 1 Enable 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 gateway 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 1 Q 850 Reason 2 SIP Reason 3 SIP
492. start at the lowest channel number in the hunt group and if that channel is not available select the next higher channel Cyclic Descending Select the next available channel in descending cycle order Always select the next lower channel number in the hunt group When the gateway reaches the lowest channel number in the hunt group it selects the highest channel number in the hunt group and then starts descending again Descending Select the highest available channel Always start at the highest channel number in the hunt group and if that channel is not available select the next lower channel Dest Number Cyclic Ascending First select the gateway port according to the called number refer to the note below If the called number isn t found then select 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 By Source Phone Number Select the gateway port according to the calling number Parameter Name in ini File Parameter Format TrunkGroupSettings Defines rules for port allocation for specific Hunt Groups If no rule exists the global rule defined by ChannelSelectMode applies TrunkGroupSettings lt Hunt group ID gt lt Channel select Mode gt lt Registration Mode gt The format is a b c Where a Hunt Group ID number b Channel select mode for that Hunt Group c Registration mode for that Hunt Group Per
493. steps Access the Embedded Web Server refer to Section 5 3 on page 51 Open the IKE Table screen Advanced Configuration menu gt Security Settings gt IKE Table option the IKE Table screen is displayed Figure 12 3 IKE Table Screen IKE Table Policy Index fo State Does not exist Internet Key Exchange table row does not exist Authentication Method Pre shared Key Shared Key a IKE SA LifeTime sec 28800 IKE SA LifeTime KB First Proposal Encryption Type Not Defined First Proposal Authentication Type Not Defined First Proposal DH Group Not Defined Second Proposal Encryption Type Not Defined Second Proposal Authentication Type Not Defined Second Proposal DH Group Third Proposal Encryption Type Not Defined Not Defined he Third Proposal Authentication Type Third Proposal DH Group Fourth Proposal Encryption Type Fourth Proposal Authentication Type Fourth Proposal DH Group Not Defined Not Defined Not Defined Not Defined 7 Not Defined z Version 5 0 283 December 2006 Ta WH wt AudioCodes MediaPack 3 In the Policy Index drop down list select the peer you want to edit up to 20 peers can be configured 4 Configure the IKE parameters according to Table 12 1 on page 281 5 Click the button Create a row is create in
494. stics Voice Compression Silence Suppression Packet Loss Concealment Echo Canceler Gain Control DTMF Transport in band DTMF Detection and Generation G 711 PCM at 64 kbps p law A law 10 20 30 40 50 60 80 100 120 msec G 723 1 MP MLQ at 5 3 or 6 3 kbps 30 60 90 msec G 726 at 32 kbps ADPCM 10 20 30 40 50 60 80 100 120 msec G 729 CS ACELP 8 Kbps AnnexA B_ 10 20 30 40 50 60 msec G 723 1 Annex A G 729 Annex B PCM and ADPCM Standard Silence Descriptor SID with Proprietary Voice Activity Detection VAD and Comfort Noise Generation CNG G 711 appendix 1 G 723 1 G 729 a b G 165 and G 168 2000 64 msec Programmable Mute transfer in RTP payload or relay in compliance with RFC 2833 Dynamic range 0 to 25 dBm compliant with TIA 464B and Bellcore TR NWT 000506 SIP User s Manual 336 Document LTRT 65408 SIP User s Manual 16 Selected Technical Specifications Table 16 2 MP 124 Functional Specifications continues on pages 336 to 338 Call Progress Tone Detection and Generation Output Gain Control Input Gain Control Fax Modem Relay Fax Relay Modem Transparency Protocols VoIP Signaling Protocol Communication Protocols Line Signaling Protocols Processor Control Processor Control Processor Memory Signal Processors Interfaces FXS Telephony Interface Network Interface RS 232 Interface Indicators Connectors amp Switches Rear Panel 24 Analo
495. sytone pcm e cpt_recorded_ringtone pcm e cpt_recorded_invalidtone pcm If the gateway is configured correctly with a Call Progress Tones dat file loaded to the gateway all four Call Progress Tones are detected by the gateway By noting whether the gateway detects the tones or not you can determine how well the Call Progress Tones dat file matches your PBX During the first run of the CPTWizard it is likely that the gateway does not detect any tones Some tones cannot be detected by the MediaPack gateway hardware such as 3 frequency tones and complex cadences CPTWizard is therefore limited to detecting only those tones that can be detected on the MediaPack gateway At this stage you can either click Next to generate a Call Progress Tones ini file and terminate the wizard or continue to manual recording mode SIP User s Manual 370 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools E 2 5 Recording Screen Manual Mode In manual mode you can record and analyze tones included in the Call Progress Tones ini file in addition to those tones analyzed when in automatic mode gt To start recording in manual mode take these 6 steps 1 Click the Manual tab at the top of the recording screen the manual recording screen is displayed Figure E 9 Recording Screen Manual Mode E AudioCodes Call Progress Tones Wizard 2 x Automatic Manual m Manual tone recording and analysis lt
496. t of the Request URI for IP gt Tel calls and sets the Contact header to the source number for Tel gt IP calls default 1 Sets the destination number to the user part of the To header for IP gt Tel calls and sets the Contact header to the username parameter for Tel IP calls 0 The gateway doesn t require client certificate default 1 The gateway when acting as a server for the TLS connection requires reception of client certificate to establish the TLS connection Note The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName 0 Use 182 Queued response to indicate a call waiting default 1 Use 180 Ringing response to indicate a call waiting Defines the maximum number of active SIP dialogs that are not call related i e REGISTER and SUBSCRIBE The valid range is 1 to 5 The default value is 5 Enables Japan NTT Modem Direct Inward Dialing DID support FXS gateways 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 applicable to FXS gateways The DID signal can be sent alone or combined with an NTT Caller ID signal Enables generation of Japan NTT Modem DID signal per port The format is EnableDID_ lt Port gt lt Modem DID gt Modem DID 0 Disabled default 1 Enabled If not configured use the global parameter EnableDID Note Appli
497. t only if the Ethernet connection is full duplex Note EnableLanWatchDog is not applicable to MP 118 Version 5 0 81 December 2006 Ta fa AudioCodes MediaPack Table 5 8 Advanced Parameters General Parameters continues on pages 78 to 82 Parameter Enable Calls Cut Through CutThrough EnableUserlnfoUsage Enable User Information Usage FXSOOSBehavior Out Of Service Behavior Description Enables users to receive incoming IP calls while the port is in an offhooked state Disable 0 Disabled default Enable 1 Enabled If enabled FXS gateways answer the call and cut through the voice channel if there is no other active call on that port even if the port is in offhooked state When the call is terminated by the remote party the gateway plays a reorder tone for TimeForReorderTone seconds and is then ready to answer the next incoming call without onhooking the phone The waiting call is automatically answered by the gateway when the current call is terminated EnableCallWaiting 1 Note This option is applicable only to FXS gateways Enables or disables usage of the User Information loaded to the gateway via the User Information auxiliary file 0 Disable default 1 Enable Determines the behavior of FXS endpoints that are not defined in the Endpoint Phone Number table and the behavior of all FXS endpoints when a Busy Out condition exists 0 None Normal opera
498. t was directly connected to the remote host on a point to point connection Since PPPoE frames are sent over Ethernet each PPP session must learn the Ethernet address of the remote peer as well as establish a unique session identifier The PPPoE standard describes a discovery protocol that provides this A PPPoE session begins with a discovery phase Only after this discovery is completed can the PPP state machine start with LCP Authentication etc as described above Each of the Ethernet frames carrying PPP session has a standard Ethernet header followed by PPPoE header and is sent with the remote host Ethernet MAC address except for the very first one in the discovery phase which is broadcasted to all hosts Further information on the transmission of PPPoE is available on the IETF website http www ietf org ric rfc2516 txt PPPoE in AudioCodes Gateways The AudioCodes gateway contains a PPPoE client embedded in its software When correctly configured see ini file parameters the gateway can try to connect to a remote PPPoE Access Concentrator When restarting the gateway after several BOOTP attempts if PPPoE is enabled see ini file parameter EnablePPPoE the gateway tries to initiate a PPP session The gateway initiates a PPPoE discovery phase to discover a PPPoE Access Concentrator It does this by broadcasting a discovery initialization packet PADI If an Access Concentrator exists and replies the gateway tries to conn
499. tained in the ifTable directory 14 1 3 SNMP Extensibility Feature One of the principal components of an SNMP manager is a MIB Compiler which allows new MIB objects to be added to the management system When a MIB is compiled into an SNMP manager the manager is made aware of new objects that are supported by agents on the network The concept is similar to adding a new schema to a database SIP User s Manual 306 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management Typically when a MIB is compiled into the system the manager creates new folders or directories that correspond to the objects These folders or directories can typically be viewed with a MIB Browser which is a traditional SNMP management tool incorporated into virtually all Network Management Systems The act of compiling the MIB allows the manager to know about the special objects supported by the agent and access these objects as part of the standard object set 14 2 Carrier Grade Alarm System The basic alarm system has been extended to a carrier grade alarm system A carrier grade alarm system provides a reliable alarm reporting mechanism that takes into account EMS outages network outages and transport mechanism such as SNMP over UDP A carrier grade alarm system is characterized by the following m The device has a mechanism that allows a manager to determine which alarms are currently active in the device That is the device maintains an
500. talian Latvian Lithuanian Maltese Norwegian Polish Portuguese Slovak Slovene Spanish Swedish Alulirott AudioCodes Ltd nyilatkozom hogy a MP 11x FXS amp FXO Series amp MP 124 megfelel a vonatkoz alapvet6 k vetelm nyeknek s az 89 336 EEC 73 23 EEC 1999 5 ES ir nyelv egy b eldirasainak ki etta er samr mi vi tilskipun Evr6pusambandsins 89 336 EEC 73 23 EEC 1999 5 ES Con la presente AudioCodes Ltd dichiara che questo MP 11x FXS amp FXO Series amp MP 124 conforme ai requisiti essenziali ed alle altre disposizioni pertinenti stabilite dalla direttiva 89 336 EEC 73 23 EEC 1999 5 ES Ar o AudioCodes Ltd deklar ka MP 11x FXS amp FXO Series amp MP 124 atbilst Direkt vas 89 336 EEC 73 23 EEC 1999 5 ES b tiskaj m pras b m un citiem ar to saist tajiem noteikumiem AudioCodes Ltd deklaruoja kad irenginys MP 11x FXS amp FXO Series amp MP 124 tenkina 89 336 EEC 73 23 EEC 1999 5 ES Direktyvos esminius reikalavimus ir kitas sios direktyvos nuostatas Hawnhekk AudioCodes Ltd jiddikjara li dan MP 11x FXS amp FXO Series amp MP 124 jikkonforma mal ti ijiet essenzjali u ma provvedimenti o rajn relevanti li hemm fid Dirrettiva 89 336 EEC 73 23 EEC 1999 5 ES Dette produktet er i samh righet med det Europeiske Direktiv 89 336 EEC 73 23 EEC 1999 5 ES AudioCodes Ltd deklarujemy z pelna odpowiedzialnoscia ze wyr b MP 11x FXS amp FXO Series amp MP 124
501. te 1 DTMF negotiation methods are prioritized according to the order of their appearance Note 2 When out of band DTMF transfer is used 1 2 or 3 the parameter DTMFTransportT ype is automatically set to 0 DTMF digits are erased from the RTP stream Note 3 When RFC 2833 4 is selected the gateway Negotiates RFC 2833 Payload Type PT using local and remote SDPs Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP Expects to receive RFC 2833 packets with the same PT as configured by the parameter RFC2833PayloadType Uses the same PT for send and receive if the remote party doesn t include the RFC 2833 DTMF PT in its SDP Note 4 When TxDTMFOption is set to 0 the RFC 2833 PT is set according to the parameter RFC2833PayloadType for both transmit and receive ini file note The DTMF transmit methods are defined using a repetition of the same TxDTMFOption parameter up to five options can be provided The RFC 2833 DTMF relay dynamic payload type Range 96 to 99 106 to 127 Default 96 The 100 102 to 105 range is allocated for proprietary usage Note 1 Cisco is using payload type 101 for RFC 2833 Note 2 When RFC 2833 payload type PT negotiation is used TxDTMFOption 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 Supported hook flash Transport Type method by which hook flash is
502. ted on a desktop or installed in a standard 19 inch rack Refer to Section 3 2 4 on page 38 for cabling the MP 124 Mounting the MP 124 on a Desktop No brackets are required Simply place the MP 124 on the desktop in the position you require Figure 3 9 Desktop or Shelf Mounting SIP User s Manual 36 Document LTRT 65408 SIP User s Manual 3 Installing the MediaPack Rack Mount Safety Instructions UL When installing the chassis in a rack be sure to implement the following Safety instructions recommended by Underwriters Laboratories e Elevated Operating Ambient If installed in a closed or multi unit rack assembly the operating ambient temperature of the rack environment may be greater than room ambient Therefore consideration should be given to installing the equipment in an environment compatible with the maximum ambient temperature Tma specified by the manufacturer Reduced Air Flow Installation of the equipment in a rack should be such that the amount of air flow required for safe operation on the equipment is not compromised Mechanical Loading Mounting of the equipment in the rack should be such that a hazardous condition is not achieved due to uneven mechanical loading Circuit Overloading Consideration should be given to the connection of the equipment to the supply circuit and the effect that overloading of the circuits might have on overcurrent protection and supply wiring Appropriate considerat
503. ters for the signal processor of the loop interface devices This parameter set provides control of 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 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 digital design of the filters and gain stages also ensures high reliability no drifts over temperature or time and simple variations between different line types In future software releases it is to be expanded to consist of different sets of line parameters which can be selected in the ini file for each port Note Use the parameter CountryCoefficients described in Table 5 35 on page 132 to configure the FXO coefficients Version 5 0 331 December 2006 Ta WH wi AudioCodes MediaPack 15 4 User Information File The User Information file maps PBX extensions connected to the MediaPack gateway to global IP phone numbers alphanumerical In this context a global IP number 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 that the mapping mechanism is disabled by
504. th a gateway without needing to power cycle the gateway As with any BootP session the computer running the BootP Tool must be located on the same subnet as the controlled VoIP gateway Log Window The log window refer to Figure C 1 on the previous page records all BootP request and BootP reply transactions as well as TFTP transactions For each transaction the log window displays the following information Client shows the Client address of the VoIP gateway which is the MAC address of the client for BootP transactions or the IP address of the client for TFTP transactions Date shows the date of the transaction based on the internal calendar of the computer Time shows the time of day of the transaction based on the internal clock of the computer Status indicates the status of the transaction e Client Not Found A BootRequest was received but there is no matching client entry in the BootP Tool e Client Found A BootRequest was received and there is a matching client entry in the BootP Tool A BootReply is sent e Client s MAC Changed There is a client entered for this IP address but with a different MAC address e Client Disabled A BootRequest was received and there is a matching client entry in the BootP tool but this entry is disabled e Listed At Another BootP utility is listed as supporting a particular client when the Test Selected Client button is clicked for details on Testing a client refer to Section C
505. th blank parameters is displayed 2 Enter values in the fields on the right side of the window using the guidelines for the fields in Section C 11 5 on page 357 3 Click Apply to save this entry to the list of clients or click Apply amp Reset to save this entry to the list of clients and send a reset message to that gateway to immediately implement the settings Note To use Apply amp Reset you must enable ARP Manipulation in the Preferences window Also you must have administrator privileges for the computer you are using An easy way to create several clients that use similar settings is to create a template For information on how to create a template refer to Section C 12 on page 360 Version 5 0 355 December 2006 Ta WH wt AudioCodes MediaPack gt To add a client to the list using a template take these 5 steps Click Add New Client gy a client with blank parameters is displayed 2 In the field Template located on the right side of the Client Configuration Window click on the down arrow to the right of the entry field and select the template that you want to use 3 The values provided by the template are automatically entered into the parameter fields on the right side of the Client Configuration Window To use the template parameters leave the check box next to that parameter selected The parameter values appear in gray text 4 To change a parameter to a different value unselect the check
506. thProtocol Authentication protocol to be used for the SNMP v3 user SNMPUsers_AuthProtocol 0 none default 1 MD5 2 SHA 1 PrivProtocol Privacy protocol to be used for the SNMP v3 user SNMPUsers_PrivProtocol 0 none default 1 DES 2 3DES 3 AES128 4 AES192 5 AES256 AuthKey Authentication key Keys can be entered in the form of a text SNMPUsers_AuthKey password or long hex string Keys are always persisted as long hex strings and keys are localized PrivKey Privacy key Keys can be entered in the form of a text password or SNMPUsers_PrivKey long hex string Keys are always persisted as long hex strings and keys are localized Group The group with which the SNMP v3 user is associated SNMPUsers_Group 0 read only group default 1 read write group 2 trap group Note all groups can be used to send traps Version 5 0 181 December 2006 fa AudioCodes MediaPack 5 6 6 4 Advanced Configuration ini File Parameters Table 5 55 describes the board parameters that can only be configured via the ini file Table 5 55 Board ini File Parameters continues on pages 182 to 184 ini File Parameter Name LifeLineType EnableDiagnostics EnableParametersMonitoring WatchDogStatus DisableRS232 DisableWebTask ResetWebPassword DisableWebConfig HTTPport HeartBeatDestIP HeartBeatDestPort HeartBeatIntervalmsec Valid Range and Description The Lifeline is activated on 0 Power down defa
507. the File Data window press the Esc key to cancel your changes you are returned to the Prerecorded Tones File s screen Figure E 5 File Data Window File Data x ooe i Code z Description Default msec 6 In the Output field specify the output directory in which the PRT file is generated followed by the name of the PRT file the default name is prerecordedtones dat Alternatively use the Browse button to select a different output file Navigate to the desired file and select it the selected file name and its path appear in the Output field 7 Click the Make File s button the Progress bar at the bottom of the window is activated The dat file is generated and placed in the directory specified in the Output field A message box informing you that the operation was successful indicates that the process is completed E 2 Call Progress Tones Wizard This section describes the Call Progress Tones Wizard CPTWizard an application designed to facilitate the provisioning of a MediaPack FXO gateway by recording and analyzing Call Progress Tones generated by any PBX or telephone network E 2 1 About the Call Progress Tones Wizard The Call Progress Tones wizard helps detect the Call Progress Tones generated by your PBX or telephone exchange and creates a basic Call Progress Tones ini file containing definitions for all relevant Call Progress Tones providing a good starting point when configuring
508. the AOR for the user by stripping the parameter The resulting URI is the AOR For example AOR sip alice example com GRUU sip alice example com opaque kjh29x97us97d Disable 0 Tgrp parameter isn t used default Send Only 1 The hunt group number is added as the tgrp parameter to 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 Send and Receive 2 The functionality of outgoing SIP messages is identical to the functionality described in option 1 In addition for incoming SIP messages if the Request URI includes a tgrp parameter the gateway routes the call according to that value if possible If the Contact header includes a tgrp parameter it is copied to the corresponding outgoing messages in that dialog Defines the string that is used in the SIP request header User Agent and SIP response header Server If not configured the default string AudioCodes product name s w version is used e g User Agent Audiocodes Sip Gateway MP 118 FXS v 4 80 004 008 When configured the string UserAgentDisplaylnfo s w version is used e g User Agent MyNewOEM v 4 80 004 008 Note that the version number can t be modified The maximum string length is 50 characters Defines the value of the Subject header in outgoing INVITE
509. the IKE table 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 To delete a peer from the IKE table select it in the Policy Index drop down list click the button Delete and click OK in the prompt 12 1 3 2 IPSec Configuration The parameters described in Table 12 3 below are used to configure the SPD table A different set of parameters can be configured for each of the 20 available IP destinations Table 12 3 SPD Table Configuration Parameters continues on pages 284 to 285 Parameter Name Description Remote IP Address Defines the destination IP address or a FQDN the IPSecPolicyRemotelPAddres IPSec mechanism is applied to s This parameter is mandatory Note When a FQDN is used a DNS server must be configured DNSPriServerlP Local IP Address Type Determines the local interface to which the encryption is IPSecPolicyLocallPAddressT applied applicable to multiple IPs and VLANs ype 0 OAM interface default IPSec is applied to 1 Control interface outgoing packets whose IP address Source Port Defines the source port the IPSec mechanism is rati IPSecPolicySrcPort applied to Serle au The default value is 0 any port protocol type match Destination Port Defines the destination port the IPSec mechanism is the values defined for IPSecPolicyDstPort applied to these four The default value is 0 any port parameters Protocol Defines the protocol type
510. the MAC address of an existing client is edited a new client is added with the same parameters as the previous client Client Name Enter a descriptive name for this client so that it is easier to remember which VoIP gateway the record refers to For example this name could refer to the location of the gateway Template Click the pull down arrow if you wish to use one of the templates that you configured This applies the parameters from that template to the remaining fields Parameter values that are applied by the template are indicated by a check mark in the box to the right of that parameter Uncheck this box if you want to enter a different value If templates are not used the box to the right of the parameters is colored gray and is not selectable IP Enter the IP address you want to apply to the VoIP gateway Use the normal dotted decimal format Subnet Enter the subnet mask you want to apply to the VoIP gateway Use the normal dotted decimal format Ensure that the subnet mask is correct If the address is incorrect the VoIP gateway may not function until the entry is corrected and a BootP reset is applied Gateway Enter the IP address for the data network gateway used on this subnet that you want to apply to the VoIP gateway The data network gateway is a device such as a router that is used in the data network to interface this subnet to the rest of the enterprise network TFTP Server IP This field contains the IP address
511. the Proxy server the gateway currently operates with Current Proxy 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 5 0 191 December 2006 Ta WH wt AudioCodes MediaPack 5 7 2 Activating the Internal Syslog Viewer The Message Log screen displays Syslog debug messages sent by the gateway Note that it is not recommended to keep a Message Log session open for a prolonged period refer to the Note below For prolong debugging use an external Syslog server refer to Section 13 2 on page 301 Refer to the Debug Level parameter GwDebugLevel described in Table 5 8 to determine the Syslog logging level gt 1 Log is Acti 210 23h 48m 23s 21da 23h 48m 23s 21da 23h 48m 23s 21da 23h 48m 23s 21da 23h 48m 23s To activate the Message Log take these 4 steps In the General Parameters screen under Advanced Parameters submenu accessed from the Protocol Management menu set the parameter Debug Level to 5 This parameter determines the Syslog logging level in the range 0 to 5 where 5 is the highest level Open the Message Log screen Status amp Diagnostics menu gt Message Log the Message Log screen is displayed and the Log is activated Figure 5 58 Message Log Screen vated lgr_flow 380 0 0FF_HOOK EV lgr_flow 381 0 O0FF_ HOOK EV lgr_psbrdif 38
512. the gateway s default coders refer to Section 5 5 1 3 on page 72 or one of the coder groups you defined in the Coder Group Settings screen refer to Section 5 5 6 1 on page 108 Repeat steps 2 to 6 for the second to fifth IP Profiles optional Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 24 ini File IP Profile Settings Description IPProfile_ lt Profile ID gt lt Profile Name gt lt Preference gt lt Coder Group ID gt lt IsFaxUsed gt lt DJBufMinDelay gt lt DJBufOptFactor gt lt IpDiffServ gt lt ControllPDiffServ gt lt N A use instead gt lt RTPRedundancyDepth gt lt RemoteBaseUDPPort gt lt CNGmode gt lt VxxTransportType gt lt NSEMode gt lt PlayRBTone2IP gt lt EnableEarlyMedia gt lt ProgressIndicator2IP gt For example IPProfile_1 name1 2 1 0 10 13 15 44 1 1 6000 0 2 0 0 1 0 IPProfile_2 name2 5 1 55 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note This parameter can appear up to 9 times ID 1 to 9 SIP User s Manual 114 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 7 Configuring the Endpoint Phone Numbers From the Endpoint Phone Number Table screen you can enable and assign telephone numbers hunt groups optional
513. the parameter NoOpInterval Determines the payload type of No Op packets The valid range is 96 to 127 The default value is 120 Changes the RTP packets according to the MAC address of received RTP packets and according to Gratuitous Address Resolution Protocol GARP messages Valid options include 0 nothing is changed 1 If the gateway 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 gateway s ARP cache table 2 The gateway uses the received GARP packets to change the MAC address of the transmitted RTP packets 3 both 1 and 2 options above are used default Version 5 0 151 December 2006 Ta A wi AudioCodes MediaPack 5 6 2 Configuring the Media Settings From the Media Settings page you can define Voice Settings refer to Section 5 6 2 1 below Fax Modem CID Settings refer to Section 5 6 2 2 on page 154 RTP RTCP Settings refer to Section 5 6 2 3 on page 157 Hook Flash Settings refer to Section 5 6 2 4 on page 160 General Media Settings refer to Section 5 6 2 5 on page 161 These parameters are applied to all MediaPack channels Several Channels Settings parameters can be configured per call using profiles refer to Section 5 5 6 on page 108 Those parameters contained within square brackets are the names used to configure the para
514. the parameter RegistrationTime No 0 Use the gateway s IP address in keep alive OPTIONS messages default Yes 1 Use GatewayName in keep alive OPTIONS messages The OPTIONS Request URI host part contains either the gateway s IP address or a string defined by the parameter Gatewayname The gateway uses the OPTIONS request as a keep alive message to its primary and redundant Proxies EnableProxyKeepAlive 1 No 0 Gateway fallback is not used default Yes 1 Internal Tel to IP Routing table is used when Proxy servers are not available When the gateway falls back to the internal Tel to IP Routing table the gateway continues scanning for a Proxy When the gateway finds an active Proxy it switches from internal routing back to Proxy routing Note To enable the redundant Proxies mechanism set EnableProxyKeepAlive to 1 or 2 Determines if the local Tel to IP routing table takes precedence over a Proxy for routing calls No 0 Only Proxy is used to route calls default Yes 1 The gateway checks the Dest Phone Prefix and or Source Phone Prefix field in the Tel to IP Routing table for a match with the outgoing call Only if a match is not found a Proxy is used Note Applicable only if Proxy is not always used AlwaysSendToProxy 0 SendInviteToProxy 0 SIP User s Manual 70 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 4 Proxy
515. tion E Outgoing packets from the gateway to the switch All outgoing packets are tagged each according to its interface control media or OAM If the gateway s native ID is identical to one of the other IDs usually to the OAM ID this ID e g OAM is set to zero on outgoing packets VlanSendNonTaggedOnNative 0 This method is called Priority Tagging p tag without Q tag If the parameter VianSendNonTaggedOnNative is set to 1 the gateway sends regular packets with no VLAN tag E Incoming packets from the switch to the gateway The switch sends all packets intended for the gateway according to the switch s configuration to the gateway without altering them For packets whose VLAN ID is identical to the switch s PVID the switch removes the tag and sends a packet The gateway only accepts packets that have a VLAN ID identical to one of its interfaces control media or OAM Packets with a VLAN ID that is 0 or packets without a tag are accepted only if the gateway s native VLAN ID is identical to the VLAN ID of one of its interfaces In this case the packets are sent to the relevant interface All other packets are rejected 9 9 3 Getting Started with VLANS and Multiple IPs By default the MediaPack operates without VLANs and multiple IPs using a single IP address subnet mask and default gateway IP address This section provides an example of the configuration required to integrate the MediaPack into a VLAN and multip
516. tion No response is provided to undefined endpoints Dial tone is played to FXS endpoints when a Busy Out condition exists 1 Reorder Tone The gateway plays a reorder tone to the connected phone PBX default 2 Polarity Reversal The gateways reverses the polarity of the endpoint marking it as unusable relevant for example to PBX DID lines This option can t be configured on the fly 3 Polarity Reversal Reorder Tone Same as 2 and 3 combined This option can t be configured on the fly SIP User s Manual 82 Document LTRT 65408 SIP User s Manual 5 Web Management 5 5 2 2 Supplementary Services Use this screen to configure parameters that are associated with supplementary services For detailed information on the supplementary services refer to Section 8 1 on page 215 gt To configure the supplementary services parameters take these 4 steps 1 Open the Supplementary Services screen Protocol Management menu gt Advanced Parameters submenu gt Supplementary Services option the Supplementary Services screen is displayed Figure 5 10 Supplementary Services Parameters Screen Supplementary Services Enable Hold Enable Hold Format 0 0 0 0 Enable Transfer Enable Transfer Prefix Enable Call Forward Enable Enable Call Waiting Enable Number of Call Waiting Indications 2 Time Between Call Waiting Indications 10 T
517. tion phone number for Tel gt IP calls Note 1 This option can be used to define various routing rules Note 2 To use this feature you must configure the hunt group IDs No 0 Disable the add port as prefix service default Yes 1 Enable the add port as prefix service If enabled then the gateway s port number single digit in the range 1 to 8 for 8 port gateways two digits in the range 01 to 24 in MP 124 is added as a prefix to the destination phone number for Tel gt IP calls Note This option can be used to define various routing rules SIP User s Manual 98 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 16 Routing Tables General Parameters continues on pages 98 to 99 Parameter IP to Tel Remove Routing Table Prefix RemovePrefix Enable Alt Routing Tel to IP AltRoutingTel2IPEnable Alt Routing Tel to IP Mode AltRoutingTel2IPMode Max Allowed Packet Loss for Alt Routing IPConnQoSMaxAllowedP L Max Allowed Delay for Alt Routing msec IPConnQoSMaxAllowedD elay Description No 0 Don t remove prefix default Yes 1 Remove the prefix defined in the IP to Hunt Group Routing table from a telephone number for an P gt Tel call before forwarding it to Tel For example To route an incoming IP gt Tel Call with destination number 21100 the IP to Hunt Group Routing table is scanned for a matching prefix If such prefix is found 21 for instance the
518. tional Specifications continues on pages 336 to 338 Type Approvals Safety and EMC Management Configuration Management and Maintenance UL 60950 1 FCC part 15 Class B CE Mark EN 60950 1 EN 55022 EN 55024 EN61000 3 2 EN61000 3 3 EN55024 Gateway configuration using Web browser or ini files SNMP v2c SNMP v3 Syslog per RFC 3164 Local RS 232 terminal Web Management via HTTP or HTTPS Telnet All specifications in this document are subject to change without prior notice SIP User s Manual 338 Document LTRT 65408 SIP User s Manual A MediaPack SIP Software Kit A MediaPack SIP Software Kit Table A 1 describes the standard supplied software kit for MediaPack FXS FXO SIP gateways The supplied documentation includes this User s Manual the MP 11x amp MP 124 MGCP H 323 SIP Fast Track Guide and the MP 11x amp MP 124 SIP Release Notes File Name Ram cmp files MP124 SIP_xxx cmp MP118_SIP_xxx cmp ini files and utilities SIPgw_MP 124 ini SIPgw_fxs_MP118 ini SIPgw_fxs_MP114 ini SIPgw_fxs_MP112 ini Usa_tones_xx dat Usa_tones_xx ini MP1xx_Coeff_FXS dat DConvert exe ACSyslog08 exe bootp exe CPTWizard exe MIBs Files Table A 1 MediaPack SIP Supplied Software Kit Description Image file containing the software for the MP 124 FXS gateway Common Image file Image file containing the software for MP 11x FXS gateways Sample ni file for MP 124 FXS gateway Sample ini file for MP 118
519. tions e Yes Reset starts only after the user defined time in the Shutdown Timeout field refer to Step 4 expires or 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 SIP User s Manual 206 Document LTRT 65408 SIP User s Manual 5 Web Management 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 gateway resets Note that if no traffic exists and the time has not expired the gateway resets 8 Click the Reset button If Graceful Option is set to Yes the reset is delayed and a screen displaying the number of remaining calls and time is displayed When the device resets a message is displayed informing of the waiting period 5 11 Logging Off the Embedded Web Server The Log Off button enables you to log off the Embedded Web Server and to re access it with a different account For detailed information on the Web User Accounts refer to Section 5 2 1 on page 49 gt To log off the Embedded Web Server take these 2 steps 1 Click the Log Off button on the main menu bar the Log Off prompt screen is displayed Figure 5 74 Log off Prompt _ Microsoft Internet Explorer 2 Log Off 2 Click OK the Web session is logged off Version 5 0 207 December 2006 7 T w
520. tiple 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 MediaPack gt To configure the NFS Settings parameters take these 7 steps 1 Open the Application Settings screen Advanced Configuration menu gt Network Settings gt Application Settings option the Application Settings screen is displayed Figure 5 36 2 Open the NFS Table screen by clicking the arrow sign gt to the right of the NFS label the NFS Table screen is displayed Figure 5 37 Figure 5 37 NFS Settings Table Screen NFS Settings gt Line Nfs gt Edit N mber Host IP Root Path Vardon Auth Type UID GID Vian Type faudiot 3 AUTO UNIX Mo gfi g MEDIA fi jaudio2 3 AUTO UNIX fo gi AMEDA E 2 92 168 jbootfiles 3 AUTO UNIX Efo Mf AMEDA El 2 x 3 To add a remote NFS file system select an available line number from the Line Number drop down list 4 Click the Add an Empty Line button an empty line appears 5 Configure the NFS Settings according to Table 5 38 6 Click the Apply New Settings button the remote NFS file system is mounted immediately Check the Syslog server for the NFS mount was successful message 7 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Note To avoid terminating calls in progress a row must not be deleted or modified while the board i
521. to IP calls NumberMapTel2IP a b c d e f g a Destination number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Number Plan used in RPID header f Number Type used in RPID header g Source number prefix The b to f manipulation rules are applied if the called and calling numbers match the a and g conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example NumberMapTel2IP 01 2 972 0 0 NumberMaPTel2IP 03 2 667 0 0 22 Note Number Plan amp Type can optionally be used in Remote Party ID RPID header by using the EnableRP Header parameter Version 5 0 93 December 2006 Ta Ce AudioCodes MediaPack Table 5 14 Number Manipulation ini File Parameters continues on pages 93 to 95 Parameter NumberMapIP2Tel SourceNumberMapTel2IP Description Manipulate the destination number for IP to Tel calls NumberMapIP2Tel a b c d e f g h i a Destination number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix
522. to your requirements refer to Table 5 10 In the Tel to IP Routing table Section 5 5 5 2 on page 100 assign a charge code rule to the routing rules you require When a new call is established the Tel to IP Routing table is searched for the destination IP addresses Once a route is found the Charge Code configured for that route is used to associate the route with an entry in the Charge Codes table Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 10 Metering Tones Parameters Description Generate Metering Tones Determines the method used to configure the metering tones that are generated to PayPhoneMeteringMode the Tel side FXS gateways only Disable 0 Metering tones aren t generated default Internal Table 1 Metering tones are generated according to the internal table configured by the parameter ChargeCode RADIUS 2 N A Note This parameter is not applicable to the Metering Tones Relay mechanism described in Section 11 on page 277 Metering Tones Type Defines the metering tone 12 kHz or 16 kHz that is detected by FXO gateways MeteringType and generated by FXS gateways 12 kHz 0 12 kHz metering tone default 16 kHz 1 16 kHz metering tone Note Suitable 12 kHz or 16 KHz coeff must be used for both FXS and FXO gateways Charge Codes Table For detailed information on configuri
523. ts to leave a Prefix suffix to add Figure 5 14 on the previous page exemplifies the use of these manipulation rules in the Source Phone Number Manipulation Table for Tel gt IP Calls When destination number equals 035000 and source number equals 20155 the source number is changed to 97220155 When source number equals 1001876 it is changed to 587623 Source number 1234510012001 is changed to 20018 Source number 3122 is changed to 2312 SIP User s Manual 92 Document LTRT 65408 SIP User s Manual 5 Web Management Table 5 13 Number Manipulation Parameters continues on pages 92 to 93 Parameter Presentation Description Select Allowed to send Caller ID information when a call is made using these destination source prefixes Select Restricted if you want to restrict Caller ID information for these prefixes When set to Not Configured the privacy is determined according to the Caller ID table refer to Section 5 5 9 3 on page 121 Note If Presentation is set to Restricted and Asserted Identity Mode is set to P Asserted the From header in INVITE message is From anonymous lt sip anonymous anonymous invalid gt and privacy id header is included in the INVITE message Table 5 14 Number Manipulation ini File Parameters continues on pages 93 to 95 Parameter NumberMapTel2IP Description Manipulates the destination number for Tel
524. tself cannot be viewed via SNMP Web or any other application m inifile behavior Secret tables are never displayed in an uploaded ini file e g when performing a Get ini File from Web operation Instead there is a commented title that states that the secret table is present at the gateway and is not to be revealed Secret tables are always kept in the gateway s non volatile memory and can be over written by new tables that are provided in a new ini file If a secret table appears in an ini file it replaces the current table regardless of its content To delete a secret table from the gateway provide an empty table of the same type with no data lines as part of a new ini file the empty table replaces the previous table in the gateway SIP User s Manual 268 Document LTRT 65408 SIP User s Manual 10 Advanced System Capabilities 10 5 5 10 5 5 1 Using the ini File to Configure Parameter Tables You can use the ini file to add modify parameter tables When using tables Read Only parameters are not loaded as they cause an error when trying to reload the loaded file Therefore Read Only parameters mustn t be included in tables in the ini file Consequently tables are loaded with all parameters having at least one of the following permissions Write Create or Maintenance Write Parameter tables in an uploaded ini file are grouped according to the applications they configure e g NFS IPSec When loading an ini f
525. ty Mode Disabled v Fax Signaling Method No Fax v Detect Fax on Answer Tone Initiate T 38 on Preamble SIP Transport Type UDP v SIP UDP Local Port 5060 SIP TCP Local Port 5060 SIP TLS Local Port 5061 Enable SIPS Disable v Enable TCP Connection Reuse Enable Mj SIP Destination Port 5060 Use userphone in SIP URL Yes he Use userphone in From Header No v Use Tel URI for Asserted Identity Disable v Tel to IP No Answer Timeout 180 Enable Remote Party ID Disable v Add Number Plan and Type to Remote Party ID Header Yes Enable History Info Header Disable v Use Source Number as Display Name No he Use Display Name as Source Number No v Play Ringback Tone to IP Dont Play Play Ringback Tone to Tel Play According to Early Me Use Tgrp information Disable v Enable GRUU Disable v User Agent Information Subject Multiple Packetization Time Format None Enable Reason Header Enable Enable Semi Attended Transfer Disable 3xx Behavior Forward v SIP T1 Retransmission Timer msec 500 SIP T2 Retransmission Timer msec 4000 SIP Maximum RTX 7 Version 5 0 Dt December 2006 Ta WH wt AudioCodes MediaPack 2 Configure the general parameters under Protocol Definition according to Table 5 3 Click the Submit button to save your changes 4 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 a Table 5 3 Protocol Definition General Param
526. tyLevel M where M is either 1 noAuthNoPriv 2 authNoPriv or 3 authPriv All changes to the trap destination configuration take effect immediately gt E To delete a trap destination take the following step Remove the appropriate row from the snmpTargetAddrTable You can change the IP address and or port number for an existing trap destination The same effect can be achieved by removing a row and adding a new row gt To modify a trap destination take the following step m Modify the IP address and or port number for the appropriate row in the snmpTargetAddrTable To disable a trap destination take the following step m Change TagList on the appropriate row in the snmpTargetAddrTable to the empty string gt To enable a trap destination take the following step m Change TagList on the appropriate row in the snmpTargetAddrTable to AC_TRAP Version 5 0 321 December 2006 7a Wa wi AudioCodes MediaPack 14 9 14 10 SNMP Manager Backward Compatibility With support for the Multi Manager Trapping feature the older acSNMPManagerIP MIB object synchronized with the first index in the snmpManagers MIB table is also supported This is translated in two features SET GET to either of the two MIB objects is identical i e aS far as the SET GET are concerned OID 1 3 6 1 4 1 5003 9 10 1 1 2 7 is identical to OID 1 3 6 1 4 1 5003 9 10 1 1 2 21 1 1 3 When setting ANY IP to the acSNMPManagerlP this is the
527. u gt Management Settings the Management Settings screen is displayed Figure 5 51 2 Open the SNMP Community Strings screen by clicking the arrow sign gt to the right of the SNMP Community Strings label the SNMP Community Strings screen is displayed Figure 5 53 3 Configure the SNMP Community Strings parameters according to Table 5 53 below Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Figure 5 53 SNMP Community Strings Screen SNMP Community String Community String Access Level Public ReadOnly E ReadOnly E ReadOnly r ReadOnly ReadOnly Do Piae e i ReadWrite r f ReadWrite ReadWrite D ReadWrite ReadWrite trapus Trap Community String Note To delete a community string check the Delete checkbox to the left of the community string you want to delete and click the button Submit Table 5 53 SNMP Community Strings Parameters Parameter Description Read Only Up to five read only community strings up to 19 characters each SNMPReadOnlyCommunityString_x The default string is public Read Write Up to five read write community strings up to 19 characters each SNMPReadWriteCommunityString_x The default string is private Trap Community String Community string used in traps up to 19 c
528. uence These numbers are also used for port allocation for IP to Tel calls if the hunt group s Channel Select Mode is set to By Phone Number Version 5 0 115 December 2006 Ta fal AudioCodes MediaPack Table 5 25 Endpoint Phone Number Table continues on pages 115 to 116 Parameter Hunt Group ID Profile ID Parameter Name in ini File TrunkGroup_x ChannelList Description In each of the Hunt Group ID fields enter the hunt group ID 1 99 assigned to the channel s The same hunt group ID can be used for more than one channel and in more than one field The hunt group ID is an optional field that is used to define a group of common behavior channels 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 or channels that correspond to the hunt group ID You can configure the Hunt Group Settings table to determine the method in which new calls are assigned to channels within the hunt groups refer to Section 5 5 8 on page 117 Note If you enter a hunt group ID you must configure the IP to Hunt Group Routing Table assigns incoming IP calls to the appropriate hunt group If you do not configure the IP to Hunt Group Routing Table calls don t complete For information on how to configure this table refer to Section 5 5 5 3 Enter the number of the Tel profile that is assigned to the endpoints defined in the Channel s
529. ult 1 Power down or when link is down physical disconnect 2 Power down or when link is down or on network failure logical link disconnect Note To enable Lifeline switching on network failure LAN watch dog must be activated EnableLANWatchDog 1 Checks the correct functionality of the different hardware components on the gateway On completion of the check if the test fails the gateway sends information on the test results of each hardware component to the Syslog server 0 Rapid 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 Section 13 1 on page 301 Obsolete parameter use the parameter ActivityListToLog instead 0 Disable gateway s watch dog 1 Enable gateway s watch dog default 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 Section 4 2 4 on page 44 and to view error notification messages For information on establishing a serial communications link with the MediaPack refer to Section 10 2 on page 262 0 Enable Web management default 1 Disable Web management Resets the username and password of the primary and secondary accounts to their defaults 0 Password
530. unconditionally Busy Incoming call is forwarded if the endpoint is busy E No Reply The incoming call is forwarded if it isn t answered for a specified time On Busy or No Reply Forward incoming calls when the port is busy or when calls are not answered after a specified time Do Not Disturb Immediately reject incoming calls Upon receipt of a Do Not Disturb call the gateway responds with a 603 Decline SIP code Three forms of forwarding parties are available m Served party the party that is configured to forward the call MediaPack F XS E Originating party the party that initiated the first call MediaPack FXS or FXO Diverted party the new destination of the forwarded call MediaPack FXS or FXO The served party MediaPack FXS can be configured through the Web browser refer to Section 5 5 9 4 on page 122 or via ini file to activate one of the call forward modes These modes are configurable per gateway s endpoint Note the following SIP issues E Initiating forward When forward is initiated the gateway sends a 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 Proxy is used the proxy s IP address m Receiving forward The gateway handles 3xx responses for redirecting calls with a new contact Call Waiting The Call Waiting feature enables FXS gateways to accept an additional second call on busy end
531. urce 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 SIP User s Manual 230 Document LTRT 65408 SIP User s Manual 8 Telephony Capabilities 8 7 8 7 1 8 7 2 8 7 3 Configuring the Gateway s 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 MediaPack gateway 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 Alternative Routing Mechanism When a Tel gt IP call is routed through the Med
532. urce phone prefix gt can be single number or a range of numbers For available notations refer to Section 5 5 3 1 on page 95 Note 2 This parameter can appear up to 50 times Note 3 Parameters can be skipped by using the sign for example Prefix 10 2 10 2 202 1 Note 4 An optional IP ProfilelD 1 to 9 can be applied to each routing rule IP to Hunt Group Routing The IP to Hunt Group Routing Table is used to route incoming IP calls to groups of channels called hunt groups Calls are assigned to hunt groups according to any combination of the following three options or using each independently E Destination phone prefix m Source phone prefix m Source IP address The call is then sent to the VoIP gateway channels assigned to that hunt group The specific channel within a hunt group that is assigned to accept the call is determined according to the hunt group s channel selection mode which is defined in the Hunt Group Settings table Section 5 5 8 on page 117 or according to the global parameter ChannelSelectMode refer to Table 5 8 on page 78 Hunt groups can be used on both FXO and FXS gateways however usually they are used with FXO gateways Note When a release reason that is defined in the Reasons for Alternative IP to Tel Routing table is received for a specific P gt Tel call an alternative hunt group for that call is available To associate an alternative hunt group to an incoming IP call assig
533. ust be only one row in the table with a Host IP of 192 168 1 1 and Root Path of audio NFS Version NFSServers_NfsVersion NFS version to use with the remote file system 2 or 3 default Identifies the authentication method used with the remote file system AUTH_NULL 0 AUTH_UNIX 1 default Auth Type NFSServers_AuthType UID User ID used in authentication if using AUTH_UNIX NFSServers_UID The valid range is 0 to 65537 The default is 0 GID Group ID used in authentication if using AUTH_UNIX NFSServers_GID The valid range is 0 to 65537 The default is 1 The VLAN OAM 0 or Media 1 to use when accessing the remote VLAN Type file system The default is to use the media VLAN NFSServers_VlanType This parameter applies only if VLANs are enabled or if Multiple IPs is configured refer to Section 9 8 on page 253 Figure 5 38 below shows an example of an NFS table definition via ini file using parameter tables for information on parameter tables refer to Section 10 5 on page 267 Figure 5 38 NFS ini File Example NFSServers FORMAT NFSServers_Index NFSServers_HostOrIP NFSServers_RootPath NFSServers_NfsVersion NFSServers_AuthType NFSServers_UID NFSServers_GID NFSServers_VlanType NESServers i IO ils ewebicil s I i ip NFSServers SIP User s Manual 144 Document LTRT 65408 SIP User s Manual 5 Web Management 5 6 1 4 Configuring the IP Routing Table The IP routi
534. uthentication CDR over RADIUS isn t supported Use RADIUS for Web Telnet Uses RADIUS queries for Web and Telnet interface authentication Login Disable 0 default WebRADIUSLogin Enable 1 When enabled logging to the gateway s Web and Telnet embedded servers is performed via a RADIUS server The gateway contacts a predefined server and verifies the given username and password pair against a remote database in a secure manner Note 1 The parameter EnableRADIUS must be set to 1 Note 2 RADIUS authentication requires HTTP basic authentication meaning the username and password are transmitted in clear text over the network Therefore users are recommended to set the parameter HttpsOnly T to force the use of HTTPS since the transport is encrypted RADIUS Authentication Server IP IP address of the RADIUS authentication server Address RADIUSAuthServerlIP RADIUS Authentication Server Port number of the RADIUS authentication server Port The default value is 1645 RADIUSAuthPort RADIUS Shared Secret Secret used to authenticate the gateway to the RADIUS server Should be a SharedSecret cryptographically strong password RADIUS Authentication Settings Default Access Level Defines the default access level for the gateway when the RADIUS DefaultAccessLevel authentication response doesn t include an access level attribute The valid range is 0 to 255 The default value is 200 Security Administrator
535. vated only when the gateway is loaded from the flash memory Therefore when using BootP Load an ini file with VianMode 1 and SaveConfiguration 1 Then after the gateway is active reset the gateway with TFTP disabled or by using any method except for BootP The gateway must be connected to a VLAN aware switch and the switch s PVID must be equal to the gateway s native VLAN ID For information on how to configure VLAN parameters refer to Section 5 6 1 5 on page 147 Table 9 1 Traffic Network Types and Priority continues on pages 255 to 255 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 ca by the initiator of the ARP listener Determined by the initiator of the request Network SNMP Traps Management Bronze DNS client EnableDNSasOAM Network Depends on the traffic type NTP EnableNTPasOAM Control Premium control Management Bronze NFS oo cal Version 5 0 255 December 2006 a WH wt AudioCodes MediaPack 9 9 2 1 Opera
536. ve information that is required for the functioning of the MediaPack It is loaded to or retrieved from the device via TFTP or HTTP These protocols are unsecured and vulnerable to potential hackers Therefore an encoded ini file significantly reduces these threats You can choose to load an encoded ini file to the MediaPack When you load an encoded ini file the retrieved ini file is also encoded Use the TrunkPack Downloadable Conversion Utility to encode or decode the ini file before you load it to or retrieve it from the device Note that the encoded ini file s loading procedure is identical to the regular ini file s loading procedure For information on encoding decoding an ini file refer to Section E 1 2 on page 365 6 2 Modifying an ini File gt To modify the ini file take these 3 steps 1 Get the ini file from the gateway using the Embedded Web Server refer to Section 5 6 3 on page 165 2 Open the file the file is open in Notepad or a Customer defined text file editor and modify the ini file parameters according to your requirements save and close the file 3 Load the modified ini file to the gateway using either BootP TFTP utility or the Embedded Web Server This method preserves the programming that already exists in the device including special default values that were preconfigured when the unit was manufactured Before loading the ini file to the gateway verify that the extension of the ini file sa
537. ved on your PC is correct Verify that the check box Hide file extension for known file types My computer gt Tools gt Folder Options gt View is unchecked Then confirm that the ini file name extension is xxx ini and NOT erroneously xxx ini ini or xxx ini Version 5 0 209 December 2006 7a VT e ett AudioCodes MediaPack 6 3 The ini File Structure The ini file can contain any number of parameters The parameters are divided into groups by their functionality The general form of the ini file is shown in Figure 6 1 below Figure 6 1 ini File Structure Sub Section Name Parameter Name Parameter Name Parameter Value Parameter Value REMARK Sub Section Name 6 3 1 The ini File Structure Rules E The ini file name mustn t include hyphens or spaces use underscore instead E Lines beginning with a semi colon as the first character are ignored m A Carriage Return must be the final character of each line E The number of spaces before and after is not relevant m If there is a syntax error in the parameter name the value is ignored m Syntax errors in the parameter value field can cause unexpected errors because parameters may be set to the wrong values E Sub section names are optional E String parameters representing file names for example CallProgressTonesFileName must be placed between two inverted commas m The parameter name is NOT case se
538. ved with the software kit to set them to their correct values gt 9 To restore the networking parameters of the MP 11x to their initial state take these 4 steps Press in the Reset button uninterruptedly for a duration of more than six seconds the gateway is restored to its factory settings username Admin password Admin Assign the MP 11x IP address refer to Section 4 2 on page 41 Load your previously backed up ini file or the default ini file received with the software kit To load the ini file via the Embedded Web Server refer to Section 5 6 3 on page 165 Press again on the Reset button this time for a short period To restore the networking parameters of the MP 124 to their initial state take these 6 steps Disconnect the MP 124 from the power and network cables Reconnect the power cable the gateway is powered up After approximately 45 seconds the Ready LED turns to green and the Control LED blinks for about 3 seconds While the Control LED is blinking press shortly on the reset button located on the left side of the front panel the gateway resets a second time and is restored with factory default parameters username Admin password Admin Reconnect the network cable Assign the MP 124 IP address refer to Section 4 2 on page 41 Load your previously backed up ini file or the default ini file received with the software kit To load the ini file via the Embedd
539. vel that is applied to all users authenticated by the RADIUS server In the field Require Secured Web Connection HTTPS select HTTPS only It is important you use HTTPS secure Web server when connecting to the gateway over an open network since the password is transmitted in clear text Similarly for Telnet use SSL TelnetServerEnable 2 refer to Section 12 2 3 on page 289 Save the changes and reset the gateway refer to Section 5 10 3 on page 206 After reset when accessing the Web or Telnet servers use the username and password you configured in the RADIUS database The local system password is still active and can be used when the RADIUS server is down SIP User s Manual 296 Document LTRT 65408 SIP User s Manual 12 Security To configure RADIUS support on the gateway using the ini file Add the following parameters to the ini file For information on modifying the ini file refer to Section 6 2 on page 209 e EnableRADIUS 1 e WebRADIUSLogin 1 e RADIUSAuthServerlP IP address of RADIUS server e RADIUSAuthPort port number of RADIUS server usually 1812 e SharedSecret your shared secret e HTTPSOnly 1 e BehaviorUponRadiusTimeout 1 e RadiusLocalCacheMode 1 e RadiusLocalCacheTimeout 300 e RadiusVSAVendorlD your vendor s ID e RadiusVSAAccessAttribute code that indicates the access level attribute e DefaultAccessLevel default access level 0 to 200 12 5 Internal Firewall
540. ver after the 5 seconds timeout expires e Deny Access the gateway denies access to the Web and Telnet embedded servers e Verify Access Locally the gateway checks the local username and password In the field Local RADIUS Password Cache Timeout enter a time in seconds when this time expires the username and password verified by the RADIUS server becomes invalid and a username and password must be re validated with the RADIUS server In the field Local RADIUS Password Cache Mode select the gateway s mode of operation regarding the above mentioned Local RADIUS Password Cache Timer option e Reset Timer Upon Access upon each access to a Web screen the timer resets reverts to the initial value configured in the previous step e Absolute Expiry Timer when you access a Web screen the timer doesn t reset but rather continues decreasing In the field RADIUS VSA Vendor ID enter the vendor ID you configured in the RADIUS server When using the Web access level mechanism perform one of the following options e When RADIUS responses include the access level attribute In the field RADIUS VSA Access Level Attribute enter the code that indicates the access level attribute in the Vendor Specific Attributes VSA section of the received RADIUS packet e When RADIUS responses don t include the access level attribute In the field Default Access Level enter the default access le
541. via TFTP Version 5 0 211 December 2006 Ta WH wt AudioCodes MediaPack 7 2 Using DHCP When the gateway is configured to use DHCP DHCPEnable 1 it attempts to contact the local DHCP server to obtain the networking parameters IP address subnet mask default gateway primary secondary DNS server and two SIP server addresses These network parameters have a time limit After the time limit expires the gateway must renew its lease from the DHCP server Note that if the DHCP server denies the use of the gateway s current IP address and specifies a different IP address according to RFC 1541 the gateway must change its networking parameters If this happens while calls are in progress they are not automatically rerouted to the new network address since this function is beyond the scope of a VoIP gateway Therefore administrators are advised to configure DHCP servers to allow renewal of IP addresses Note If the gateway s network cable is disconnected and reconnected a DHCP renewal is performed to verify that the gateway is still connected to the same network When DHCP is enabled the gateway also includes its product name e g MP 118 FXS in the DHCP option 60 Vendor Class Identifier The DHCP server can use this product name to assign an IP address accordingly After power up the gateway performs two distinct DHCP sequences Only in the second sequence DHCP option 60 is co
542. vidual parameter From the Coder Group drop down list select the coder group you want to assign to that Profile You can select the gateway s default coders refer to Section 5 5 1 3 on page 72 or one of the coder groups you defined in the Coder Group Settings screen refer to Section 5 5 6 1 on page 108 Repeat steps 2 to 6 for the second to fifth Tel Profiles optional Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 23 ini File Tel Profile Settings Description TelProfile_ lt Profile ID gt lt Profile Name gt lt Preference gt lt Coder Group ID gt lt IsFaxUsed gt lt DJBufMinDelay gt lt DJBufOptFactor gt lt IPDiffServ gt lt ControllPDiffServ gt lt DtmfVolume gt lt InputGain gt lt VoiceVolume gt lt EnableReversePolarity gt lt EnableCurrentDisconnect gt lt EnableDigitDelivery gt lt ECE gt lt MWIAnalogLamp gt lt MWIDisplay gt lt FlashHookPeriod gt lt EnableEarlyMedia gt lt ProgressIndicator2IP gt For examples TelProfile_1 FaxProfile 1 1 1 40 13 22 33 0 0 0 1 0 0 0 TelProfile_2 ModemProfile 2 2 0 40 13 0 0 0 0 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note This parameter can appear up to 9 times ID 1 to 9 SIP User s Manual
543. w opens the Client Configuration window If the MAC address indicated on the line exists in the client database it is highlighted If the address is not in the client database a new client is added with the MAC address filled out You can enter data in the remaining fields to create a new client entry for that client C 10 Setting the Preferences The Preferences window Figure C 3 is used to configure the BootP Tool parameters Figure C 3 Preferences Screen xi BootP Server TFTP Server J ARP Manipulation Enabled V Enabled gt Reply Type On Interface Broadcast Unicast 0 10 13 2 66 MARP Type Directory Dynamic a 7 C Static i Boot File Mask crp Number of Timed Replies INI File Mask Timeout Maximum Retransmissions C 10 1 BootP Preferences ARP is a common acronym for Address Resolution Protocol and is the method used by all Internet devices to determine the link layer address such as the Ethernet MAC address in order to route Datagrams to devices that are on the same subnet Version 5 0 353 December 2006 Ta WH wt AudioCodes MediaPack When ARP Manipulation is enabled on this screen the BootP Tool creates an ARP cache entry on your computer when it receives a BootP BootRequest from the VoIP gateway Your computer uses this information to send messages to the VoIP gateway without using ARP again This is particularly
544. way refer to Section 5 10 3 on page 206 SIP User s Manual 258 Document LTRT 65408 SIP User s Manual 9 Networking Capabilities 9 9 3 2 Integrating Using the ini File gt To integrate the MediaPack into a VLAN and multiple IPs network using the ini file take these 3 steps 1 Prepare an ini file with parameters shown in Figure 6 1 refer to the following notes If the BootP TFTP utility and the OAM interface are located in the same network the Native VLAN ID VlanNativeVlanid must be equal to the OAM VLAN ID VlanOamVlanld which in turn must be equal to the PVID of the switch port the gateway is connected to Therefore set the PVID of the switch port to 4 in this example e Configure the OAM parameters LocalOAMPAddress LocalOAMSubnetMask and LocalOAMDefaultGW only if the OAM networking parameters are different from the networking parameters used in the Single IP Network mode e The IP Routing table is required to define static routing rules for the OAM and Control networks since a default gateway isn t supported for these networks Figure 9 5 Example of VLAN and Multiple IPs ini File Parameters VLAN Configuration VilanMode 1 VlanOamVlanId 4 VlanNativeVlanId 4 VilanControlVlanId 5 VlanMediaVlanID 6 Multiple IPs Configuration EnableMultipleIPs 1 LocalMediaIPAddress 10 33 174 50 LocalMediaSubnetMask 255 255 0 0 LocalMediaDefaultGW 10 33 0 1 LocalControlIPAddress 10 32 174 50 LocalControlSubn
545. wise the cadence is interpreted as cyclic it repeats for every ringing cycle Ring On Time specifies the duration of the ringing signal E Ring Off Time specifies the silence period of the cadence In SIP the distinctive ringing pattern is selected according to Alert Info header that is included in INVITE message For example Alert Info lt Bellcore dr2 gt or Alert Info lt http Bellcore dr2 gt dr defines ringing pattern 2 If the Alert Info header is missing the default ringing tone 0 is played 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 units for the s
546. x Source IP Address Hunt Group ID Profile ID 5010 5020 oe UELS i ATCT 1 2 3 4 5 6 F 8 zl 2 3 4 5 6 In the IP to Tel Routing Mode field select the IP to Tel routing mode refer to Table 5 18 In the Routing Index drop down list select the range of entries that you want to edit up to 24 entries can be configured Configure the IP to Hunt Group Routing table according to Table 5 18 Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 Table 5 18 IP to Hunt Group Routing Table continues on pages 103 to 104 Parameter Description IP to Tel Routing Mode Route calls before manipulation 0 P gt Tel calls are routed before the number RouteModelP2Tel manipulation rules are applied default Route calls after manipulation 1 P gt Tel calls are routed after the number manipulation rules are applied Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number prefix The prefix can be 1 to 49 digits long An asterisk represents all numbers Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number prefix The prefix can be 1 to 49 digits long An asterisk represents all numbers Version 5 0 103 December 2006 Ta fal AudioCodes
547. xist IPSec table row does not exist Remote IP Address Local IP Address Type Control Source Port Destination Port Protocol Releated Key Exchange Method Index SA Life Time sec s800 SA Life Time KB BOO First Proposal Encryption Type Not Defined First Proposal Authentication Type Not Defined A Second Proposal Encryption Type Not Defined z Second Proposal Authentication Type Not Defined Third Proposal Encryption Type Not Defined Third Proposal Authentication Type Not Defined Fourth Proposal Encryption Type Not Defined bd Fourth Proposal Authentication Type Not Defined 3 In the Policy Index drop down list select the rule you want to edit up to 20 rules can be configured 4 Configure the SPD parameters according to Table 12 3 on page 284 5 Click the button Create a row is create in the SPD table 6 To save the changes so they are available after a power fail refer to Section 5 10 2 on page 205 To delete a peer from the SPD table select it in the Policy Index drop down list click the button Delete and click OK in the prompt SIP User s Manual 286 Document LTRT 65408 SIP User s Manual 12 Security 12 1 3 3 IPSec and IKE Configuration Table s Confidentiality Since the pre shared key parameter of the IKE table must remain undisclosed measures are taken by the ini file Embedded Web Server and SNMP agent to m
548. y only This MIB complements the other proprietary MIBs The acGateway MIB has the following groups e Common _ for parameters common to both SIP and H 323 e SIP for SIP parameters only e H 323 for H 323 parameters only acAlarm This is a proprietary carrier grade alarm MIB It is a simpler implementation of the notificationLogMIB and the IETF suggested alarmMIB both also supported in all MediaPack and related devices The acAlarm MIB has the following groups e ActiveAlarm straightforward single indexed table listing all currently active alarms together with their bindings the alarm bindings are defined in acAlarm acAlarmVarbinds and also in acBoard acTrap acBoardTrapDefinitions oid_1_3 6 1 4 1 5003 9 10 1 21 2 0 e acAlarmHistory straightforward single indexed table listing all recently raised alarms together with their bindings the alarm bindings are defined in acAlarm acAlarmVarbinds and also in acBoard acTrap acBoardTrapDefinitions oid_1_3 6 1 4 1 5003 9 101 21 2 0 The table size can be altered via notificationLogMIB notificationLogMIBObjects nlmConfig niImConfigGlobalEntryLimit or notificationLogMIB notificationLogMIBObjects nImConfig nImConfigLogTable nim ConfigLogEntry nlmConfigLogEntryLimit The table size can be any value from 10 to 100 and is 100 by default SIP User s Manual 310 Document LTRT 65408 SIP User s Manual 14 SNMP Based Management Note 1 The following are
549. y tone Matches PBX name ITU Canada Tone name Busy tone Matches PBX name ITU Dominica Commonweal Tone name Busy tone Matches PBX name ITU Hongkong China Tone name Busy tone Matches PBX name ITU Jamaica Tone name Busy tone Matches PBX name ITU Korea Republic of Tone name Busy tone Matches PBX name ITU Montserrat Tone name Busy tone SIP User s Manual 372 Document LTRT 65408 SIP User s Manual E Accessory Programs and Tools Information related to matches of a tones recorded with the CPTWizard s internal database The database is scanned to find one or more PBX definitions that match all recorded tones i e dial tone busy tone ringing tone reorder tone and any other manually recorded tone all match the definitions of the PBX If a match is found the entire PBX definition is reported as comments in the ini file using the same format Figure E 12 Full PBX Country Database Match ae te Some tones matched PBX country Audc US Additional database tones guessed below remove s to use CAL Aude US US Ringback tone L PROGRESS TONE 5 Freq Hz 450 Freq Hz 500 Freq Level dBm 0 Freq Level dBm 0 t Signal On Time 10msec 180 t Signal Off Time 10msec 450 Second Signal On Time 10msec 0 Second Signal Off Time 10msec 0 If a match is found in the database consider using the database s definitions instead of the rec

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