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Release Notes Version 5.0

1. Enables the use of DNS Naming Authority Pointer NAPTR and Service Record SRV queries to discover Proxy servers Valid options include 0 A Record default 1 SRV 2 NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy IP address parameter contains a domain name without port definition e g 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 2 an NAPTR query is performed If it is successful an SRV query is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy 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 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 name
2. 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 3xxBehavior Modification Added Web support 3xx Behavior 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 CalllD branch to and from tags or change them in the new initiated INVITE Valid options include 0 forward Use different call identifiers for a redirected INVITE message default 1 redirect Use the same call identifiers Version 5 0 31 December 2006 Ta Ca AudioCodes MediaPack Series Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name EnableProxyKeepAlive Enable Proxy Keep Alive ProxyIP Proxy IP Address
3. e Support RFC 3953 The Early Disposition Type for SIP e Support for RFC 3966 The tel URI for Telephone Numbers e Support RFC 4244 An Extension to SIP for Request History Information e Supports Tel URI Uniform Resource Identifier according to RFC 2806 bis e Supports ITU V 152 Procedures for supporting Voice Band Data over IP Networks SIP Release Notes 38 Document LTRT 65608 SIP Release Notes 2 SIP Compatibility 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 payload type negotiation DTMF out of band transfer using gt INFO method lt draft choudhuri sip info digit 00 txt gt gt INFO method compatible with Cisco gateways gt NOTIFY method lt draft mahy sipping signaled digits 01 txt gt 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 3
4. 40 50 60 Enable 1 80 100 Enable w o Adaptations 2 G 723 1 30 default 60 5 3 0 6 3 1 Always 4 Disable 0 97231 90 120 default Enable 1 G 726 10 20 default 16 0 24 1 Dynamic 0 Disable 0 9726 30 40 50 60 32 2 default 120 Enable 1 80 100 120 40 3 T 38 N A N A N A N A t38fax SIP Release Notes 26 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name NumberMapIP2Tel Description G 711A 10 20 default Always 64 Dynamic 0 N A law_VBD 30 40 50 60 120 g711AlawVbd 80 100 120 G 711U 10 20 default Always 64 Dynamic 0 N A law_VBD 30 40 50 60 120 g711UlawVbd 80 100 120 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
5. Description Modification Added Registration Mode column 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 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 Modification Additional enumeration 3 Valid options include 0 None Call Detail Recording CDR information isn t sent to the Syslog server default 1 End Call CDR information is sent to the Syslog server at end of each Call 2 Start amp End Call CDR information is sent to the Syslog server at the start and at the end of each Call 3 CDR report is sent to Syslog at connection 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 Note This parameter replaces the EnableCDR parameter Modification Added support for Call Forwarding Duration in msec of the played Stutter dial tone which indicates that Call Forwarding is enabled or that there is a waiting message s T
6. Life line testing The MediaPack now features a mechanism that performs tests on the telephone lines connected to FXS and FXO ports These tests provide various line measurements Line testing is executed via SNMP only Full Mesh Routing The MediaPack now supports a combination of FXS and FXO channels 4 FXS and 4 FXO channels Each FXS channel features a lifeline that is connected to a FXO channel channel 1 FXS to channel 5 FXO channel 2 FXS to channel 6 FXO and so on Relevant parameter LifeLineType IP Multicast Supports the reception of multicast RTP streams The gateway can join an IP multicast group in order to receive an RTP stream generated by a remote server e g a Music On Hold stream to a multicast IP address PPPoE The MediaPack can now operate as a Point to Point Protocol over Ethernet PPPoE client enabling it to be integrated in a broadband access architecture mostly in ISP networks Relevant parameters EnablePPPoE PPPoEPassword PPPoERecoverlPAddress PPPoERecoverDfgwAddress PPPoERecoverSubnetMask PPPoEServerName PPPoEStaticlPAddress PPPoEUserName The Multiple IPs mechanism can now support a dual mode separating the Media from the OAM and Control networks Internal firewall The MediaPack now accommodates an internal access list facility allowing the security administrator to define network traffic filtering rules Relevant table parameter AccessList Up to five simultaneous Te
7. MP 118 8 port MP 114 4 port and MP 112 2 port 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 the 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 under 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 10 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
8. 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 Password for the voice menu used for configuration and status To activate the menu connect an analog telephone and dial three stars followed by the password The default value is 12345 Version 5 0 67 December 2006 Ta r K AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name ActivityListToLog Activity Types to Report via Activity Log Messages Description The Activity Log mechanism enables the gateway 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 PVC Parameters Value Change Changes made on the fly to parameters AFL Auxiliary Files Loading Loading of auxiliary files e g via Certificate screen DR Device Reset Device reset via the Reset Device screen FB Flash Memory Burning Burning of files parameters to flash e g Save Configuration screen SWU Device Software Update cmp loading via the Software Upgrade Wizard ARD Access to Restricted Domains Access to Restricted Domains The following screens are restricted 1 ini parameters AdminPage 2 General
9. is equal to RegistrarName if configured The RegistrarName can be any string gt Otherwise the servername is equal to RegistrarlP either FQDN or numerical IP address if configured gt Otherwise the servername is equal to ProxyName if configured The ProxyName can be any string gt Otherwise the servername is equal to ProxyIP 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 Version 5 0 37 December 2006 Ta WH gA AudioCodes MediaPack Series 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 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
10. 4 gt lt Coder Name gt lt Ptime gt lt Rate gt lt Payload Type gt lt Silence Suppression Mode gt Coder Name arr Rate Payload Type Silence ime Suppression 10 20 G 711 A law default 30 Disable 0 g711Alaw64k 40 50 60 80 ays 64 AWays 8 Enable 1 100 120 10 20 G 711 u law default 30 Disable 0 9711Ulaw64k 40 50 60 80 ays 64 Aways g Enable 1 100 120 6 729 10 20 Enable 1 9729 PA Always 8 Always 18 Enable w o i Adaptations 2 G 723 1 30 default 5 3 0 6 3 1 Disable 0 97231 60 90 default Always 4 Enable 1 10 20 G 726 default 30 Disable 0 9726 40 50 60 80 ways 32 Always 2 Enable 1 100 120 SIP Release Notes 64 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name DisconnectOnDialTone Disconnect on Dial Tone AAAIndications AAA Indications RADIUSAccServerIP RADIUS Accounting Server IP Address RADIUSAccPort RADIUS Accounting Port RADIUSAccountingType RADIUS Accounting Type SIPSubject Subject GuardTimeBetweenCalls Guard Time Between Calls EnableUserInfoUsage Enable User Information Usage UserlInfoFileName Description 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 parame
11. A possible workaround for this issue is to set the parameter BootPRetries to 5 forcing the gateway to issue 20 BootP requests for 60 seconds A second workaround is to disable the spanning tree algorithm on the port of the external switch that is connected to the gateway When RTP packets are received after a sudden large network delay 200 to 300 msec the drift correction could take about 5 seconds During this period voice towards the TDM side is silent Static NAT is not supported for local IP calls MP 118 FXO NTT Ring Detection is not supported with or without caller ID MP 118 FXO Indian Caller ID detection is not supported MP 118 FXO Metering tone billing detection is currently not supported VLAN Pass Through mode is not supported NTT caller ID type two constraints gt The NTT standard describes the CallerID type 2 generation as a sequence of an incoming call signal C amp D DTMFs and FSK modulated data Generation of the incoming call signal remains in the responsibility of the application but C D and the FSK are generated by the supplied service The signal can be generated using the UDT signal generation mechanism gt Prior to the detection of NTT CallerlD type 2 there are 2 DTMF C and D detections which remain unscreened MP 124 rev A and MP 124 rev B do not support the following gt Long haul gt Caller ID generation gt MWI generation SIP Release Notes 48
12. Control and Media Each interface has its own IP and subnet address The Dual IP mode option allows the gateway to distinguish between only two traffic types based on IP address One of the traffic types consists of a combination of two traffic types Media and Control OAM and Control or OAM and Media while the other is whichever traffic type excluded in this combination Therefore in the Dual IP mode the same IP address is assigned to two traffic types 3 Disable LCP Echos and Link disconnection auto detection support MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now allows the user to disable the Point to Point Protocol over Ethernet PPPoE disconnection auto detection feature By default the PPPoE Client embedded on the gateway s software sends Link Configuration Protocol LCP Echo packets to the server to check that the PPPoE connection is open Some Access Concentrators don t reply to these LCP Echo requests resulting in a disconnection By disabling the LCP disconnection auto detection feature the PPPoE Client doesn t send LCP Echo packets to the server and does not detect PPPoE disconnections Relevant parameter PPPoELCPEchoEnable 4 IPSec AES support MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports Advanced Encryption Standard AES for IPSec IKE tables Relevant parameters IKEPolicyProposalEncryption_X IPSecPolicyProposalEncryption_X 5 NTT DTMF based DID support MP 112 MP 114 MP 118
13. Document LTRT 65608 SIP Release Notes 2 SIP Compatibility 2 SIP Compatibility 2 1 Supported SIP Features The MediaPack SIP main features are 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 an 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 z9hG4bRaC7AU234 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 gt The servername string is defined according to the following rules gt The servername
14. Notes 70 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Reader s Notes Version 5 0 71 December 2006 AudioCodes Analog Media Gateways amp CPE SIP MediaPack MP 124 amp MP 11x Release Notes Version 5 0 www audiocodes com THE STANDARDS INSTITUTION OF ISRAEL
15. Preference 1 20 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 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 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 1 The Tel ProfilelD can be used in the Hunt group table TrunkGroup_x parameter Note 2 Profile Name assigned to a ProfileID enabling User s to identify it intuitively and easily Note 3 This parameter can appear up to 9 times ID 1 to 9 SIP Release Notes 60 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name EnableProxyKeepAlive Enable Proxy Keep Alive ProxyKeepAliveTime Proxy Keep Alive Time RegistrationTime Registration Time UseTelURIForAsserted ID UseSourceNumberAsDisp layName Use Source Number as Display Name UseDisplayNameAsSourc eNumber Use Displa
16. Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name CoderName_ID Description 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 Note 4 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 Note 5 If the coder G 729 is selected and silence suppression is disabled for this coder 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 For example CoderName g 711Alaw64k 20 0 CoderName g 11Ulaw64k 40 CoderName g7231 90 1 1 CoderName g726 0 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
17. 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 and only Industry standard terms are used throughout this manual The symbol 0x indicates hexadecimal notation Related Documentation Document Manual Name LTRT 65407 MP 11x amp MP 124 SIP User s Manual LTRT 59803 MP 11x amp MP 124 MGCP H 323 SIP Fast Track Guide Version 5 0 5 December 2006 Ta WH wt AudioCodes MediaPack Series Notes s MediaPack refers to the MP 124 MP 118 MP 114 and MP 112 VolP gateways e MP 11x refers to the MP 118 MP 114 and MP 112 VoIP gateways SIP Release Notes 6 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 1 What s New in Release 5 0 This document uses a one row table convention to indicate for which products each feature is applicable The products that don t support the feature are shaded grayed In the example below the feature would be applicable only to MP 114 MP 118 1 1 Supported Hardware Platforms 1 1 1 New Hardware Platforms Introduced in this Release The following hardware pla
18. detailed information on the parameters of each table refer to the User s Manual 2 The following NAT traversal mechanisms were added gt The gateway now supports the Simple Traversal of UDP Through NATs STUN protocol according to RFC 3489 This mechanism enables the gateway to discover the presence and types of NATs and firewalls located between it and the public Internet It also provides the gateway with the capability to determine the public IP address allocated to the NAT This information is later embedded in outgoing SIP SDP messages and enables remote SIP user agents to reach the gateway Relevant parameters EnableSTUN STUNServerPrimaryIP STUNServerSecondaryIP NATBindingDefaultTimeout gt Toenable NAT traversal for the RTP streams RTP NO OP packets according to avt rtp noop draft are now sent This method ensures that the NAT binding remains open during RTP silence periods Relevant parameters RTPNoOpEnable RTPNoOpInterval RTPNoOpPayloadType gt Can now configure the gateway to send keep alive traps to a different UDP port Relevant parameter KeepAliveTrapPort 3 The interface for handling coders was improved You can now select the coder family packetization time rate where applicable payload type where applicable and silence suppression individually per coder Relevant parameters CoderName CoderName_ID 4 Additional parameters were added to the IP and Tel Profiles In addition the number of di
19. g711Alaw64k 20 0 CoderName_1 g711Ulaw64k 40 CoderName_1 g7231 90 1 1 CoderName_2 g726 2 0 Modification asterisk wildcard supported in IP addresses Manipulates the destination number for IP to Tel calls The format for NumberMapIP2Tel is as follows a b c d e f g h i Where 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 Not applicable set to f Not applicable set to g Source number prefix h Not applicable set to 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 wildcards The x wildcard is used to represent single digits e g 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The wildcard represents any numbe
20. gateway resends its registration request after 3600 x 70 2520 sec The default value of RegistrartionTimeDivider is 50 e 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 e Single gateway Registration or multiple Registration of all gateway endpoints e Supported methods INVITE CANCEL BYE ACK REGISTER OPTIONS INFO REFER UPDATE NOTIFY PRACK SUBSCRIBE and PUBLISH e Modifying connection parameters for an already established call re INVITE e Working with Redirect server and handling 3xx responses e Early media supporting 183 Session Progress e PRACK reliable provisional responses RFC 3262 e Call Hold and Transfer Supplementary services using REFER Refer To Referred By Replaces and NOTIFY messages e Supports RFC 3711 Secured RTP and Key Exchange according to lt draft ietf mmusic sdescriptions 12 gt e Supports RFC 3489 Simple Traversal of UDP Through NATs STUN e Supports RFC 3327 Adding Path to Supported header e Supports RFC 3581 Symmetric Response Routing e Supports RFC 3605 RTCP Attribute in SDP e Supports RFC 3326 Reason header e Supports RFC 4028 Session Timers in SIP e Supports network asserted identity and privacy RFC 3325 and RFC 3323 e Support RFC 3911 The SIP Join Header e Support RFC 3903 SIP Extension for Event State Publication
21. metering rules can be defined by repeating the parameter 25 times To associate a metering rule to an outgoing Tel to IP call use the Tel to IP Routing table Prefix 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 Each Charge Code can include from a single and up to four different time periods in a day 24 hours Each time period is composed of The end in a 24 hours format of the time period The time interval between pulses in seconds The number of pulses sent on answer The first time period always starts at midnight 00 It is mandatory that the last time period of each Charge Code ends at midnight 00 This prevents undefined time frames in a day When a new call is established the Tel to IP Routing table is searched for the destination IP address 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 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 ge
22. server The IP address of the secondary STUN server Defines the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires The valid range is 0 to 2592000 The default value is 30 Enables disables sending of NO OP packets 0 Disabled default 1 Enabled This mechanism ensures that the NAT binding remains open during RTP silence periods Determines the time interval in msec in which NO OP packets is sent in the case of silence no RTP traffic The valid range is 20 to 600000 The default value is 1000 10 seconds Determines the payload type of No Op packets The valid range is 96 to 127 The default value is 120 0 Tgrp parameter isn t used default 1 send only 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 2 send and receive 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 th
23. t supported on the Mediant 1000 Determines the gateway s mode of operation when SRTP is used EnableMediaSecurity 1 0 Prefer 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 1 Must The gateway initiates encrypted calls If negotiation of the cipher suite fails the call is terminated Incoming calls that don t include encryption information are rejected default CoderName Defines the gateway s coder list up to five coders can be configured 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 Coder Name Packetization Rate Payload Type Silence Time Suppression 10 20 G 711 A law default 30 Disable 0 g711Alaw64k 40 50 60 80 ays 64 Aways amp Enable 1 100 120 10 20 G 711 u law default 30 Disable 0 9711Ulaw64k 40 50 60 80 ays 64 Aways 0 Enable 1 100 120 Disable 0 10 20 G 729 Enable 1 9729 Pan Always 8 Always 18 Enable w o ae Adaptations 2 G 723 1 30 default 5 3 0 6 3 1 Disable 0 97231 60 90 default Aways Enable 1 10 20 G 726 default 30 Disable 0 9726 40 50 60 80 Ways 32 Aways 2 Enable 1 100 120 Version 5 0 63 December 2006 Ta Ca AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File
24. 0 ini File Web Interface Parameter Name WaitForDialTime Wait For Dial Time FXSOOSBehavior Out Of Service Behavior CountryCoefficients TelConnectCode Send Digit Pattern on Connect VoiceMenuPassword Voice Menu Password Description 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 during 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 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 operation 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 endpoi
25. 118 MP 124 FXS FXO The gateway now supports Internet Draft draft ietf sip gruu O9 Obtaining and Using Globally Routable User Agent UA URIs GRUU in SIP A Globally Routable User Agent URI GRUU is a type of URI that routes to a specific UA instance and can be reached from anywhere 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 If the Registrar Proxy supports GRUU the REGISTER responses include the gruu parameter in each Contact header field In scenarios where REGISTER is sent again after expiration of the registration the Registrar Proxy provides the same GRUU for the same Address of Record AOR and instance id The gateway includes the GRUU in any header field that 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 Relevant parameter EnableGRUU Supports Registration Mode per Hunt Group MP 112 MP 114 MP 118 MP 124 FXS FXO The Registration Mode can now be configured per Hunt Group Each Hunt Group can register as one group or each endpoint in the Hunt Group can register separately Relevant parameter TrunkGroupSettings Supports q Contact parameter MP 112 MP 1
26. 14 MP 118 MP 124 FXS FXO The gateway now supports reception of a q value in the Contact header When the gateway receives a 3xx response with multiple contacts the q value may be added per contact This q value indicates the order in which the gateway should try the contacts The gateway performs serial processing in decreasing q value order while keeping the original order in case of contacts of equal q value SIP Release Notes 14 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 20 Support for RFC 3824 Using E 164 numbers with SIP ENUM 21 22 MP 112 MP 114 MP 118 MP 124 FXS FXO The E 164 Number Mapping ENUM system uses DNS to translate certain telephone numbers e g 97239764392 into URIs e g sip johnndoe audiocodes com To use ENUM the user must enter the string ENUM as the destination IP address in the Tel2IP Routing table In this scenario the gateway sends an NAPTR request to the external DNS server to resolve the received phone number into a SIP URI The SIP URI which is included in the response is used as the Request URI in the outgoing INVITE ENUM is not used for routing purposes when Proxy is in use When Proxy is not used the INVITE is sent to the IP address that was received in the ENUM reply Handling of Transfer in Alerting State MP 112 MP 114 MP 118 MP 124 FXS FXO According to Internet Draft draft ietf sipping cc transfer 05 the transferor party initia
27. 20 g711UlawVbd 80 100 120 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 g711Ulaw64k 40 CoderName g7231 90 1 1 CoderName g726 2 0 CoderName_ID Modification Support for additional coders T 38 G 711A law_VBD and G 711U law_VBD G 726 payload type and rate modified 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 Coder Name Packetization Rate Payload Type Silence Time Suppression G 711 A law 10 20 default Always 64 Always 8 Disable 0 g711Alaw64k 30 40 50 60 Enable 1 80 100 120 G 711 p law 10 20 default Always 64 Always 0 Disable 0 9711Ulaw64k 30 40 50 60 Enable 1 80 100 120 G 729 10 20 default Always 8 Always 18 Disable 0 9729 30
28. 389 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 09 gt Obtaining and Using Globally Routable User Agent UA URIs GRUUV 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 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 2 2 Unsupported SIP Features The following SIP features are NOT supported MESSAGE method Preconditions RFC 3312 SDP Simple Capability Declaration RFC 3407 S MIME Version 5 0 39 December 2006 Ta fal AudioCodes MediaPack Series 2 3 SIP Compliance Tables The MediaPack SIP gateways comply with RFC 3261 as shown in the following sections 2 3 1 SIP Functions Function User Agent Client UAC User Agent Server UAS Proxy Server Redirect Server Registrar Server Event Publication Agent EPA Event State Compositor ESC 2 3 2 SIP Methods Method INVITE ACK BYE CANCEL REGISTER REFER NOTIFY INFO OPT
29. 6 Busy Here If an endpoint is found and it is free the gateway responds with 200 OK gt Ifthe gateway responds with 200 OK it always includes an SDP body that includes all supported coders in the response in addition it fills the Supported header with the supported capabilities and the Require header if 100rel PRACK is required by the configuration The number of total and free channels was added to a 200 OK response to an OPTIONS request The gateway uses the X Resource header in the following format X Resource telchs 5 8 mediachs 0 0 Where telchs specifies the number of free tel channels total tel channels The parameter mediachs should be ignored It is now possible to determine the format of the URI in the P Asserted and P Preferred headers Relevant parameters UseTelURIForAssertedID Version 5 0 57 December 2006 Ta WH gA AudioCodes MediaPack Series 15 16 It is now possible to indicate that a call is waiting using 180 Ringing response instead of 182 Relevant parameters Send180ForCallWaiting It is now possible to configure the gateway s behavior in response to 3xx messages The redirected INVITE can either use the same call identifiers CalllD branch to and from tags or modify these identifiers as a new call Relevant Parameters 3xxBehavior 4 4 Web and SNMP New Features 1 To prevent unauthorized access to the Embedded Web Server two user accounts are no
30. Description Modification Note 3 for Proxy Keep Alive responses to INVITE requests Valid options include 0 Disable Disable default 1 Using OPTIONS Enable Keep alive with Proxy using OPTIONS 2 Using REGISTER 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 Modification EnableProxySRVQuery changed to ProxyDNSQueryT ype Added NAPTR support 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
31. Document LTRT 65608 SIP Release Notes 3 Known Constraints 20 21 22 23 24 25 26 The level field in the detection event of burst tone should be ignored is always equal to 63 dBm Setting the V 21 Transport Type to Bypass and Fax Transport Type to relay results in entering Fax Relay mode at the 2100 Hz signal Only at the end of this signal does the channel enter Bypass mode If PCM LoopBack is activated there is no way of knowing if a new channel being opened is in LoopBack state or not The parameter should only be used for test purposes When using FaxTransportType TransparentWithEvents the Fax events parameters regarding the side of the fax call answering or calling and the number of pages are invalid The number of channels operating in internal IP loopback mode that can be supported by the board is usually less than the declared channel capacity The resolution of the duration of digits On and Off time when dialing to the network using RFC 2833 relay is dependent on the basic frame size of the coder being used Flash burning control for specific files BurnCASFile BurnCallProgressToneFile BurnVXMLFile BurnVoicePromptsFile is no longer supported Everything is now controlled by the new parameter SaveConfiguration 3 4 Web Constraints 1 10 The AGC parameters in the Channel Settings screen are not applicable to the gateway Incorrect presentation of dynamic payload in
32. FXONumberOfRings SIP Release Notes 10 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 1 3 SIP New Features 1 Support for RFC 3605 RTCP attribute in SDP MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports RFC 3605 RTCP attribute in SDP When a session requires multiple ports SDP assumes that these ports have consecutive numbers RTCP Port RTP Port 1 However when the session crosses a NAT device that also uses port mapping the order of ports can change by the translation To handle this scenario an extension attribute rtcp was added The rtcp attribute documents the RTCP port used for media stream when the port is not the next higher odd port number following the RTP port described in the media line The rtcp attribute can include the RTCP port and a different IP address if relevant Support for reception of 3xx SIP messages in response to REGISTER requests MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports reception of 3xx SIP messages in response to REGISTER requests Upon receiving a 3xx response for a REGISTER request the Re REGISTER is immediately sent to the new location according to the Contact header in the 3xx response If the response is a 301 Moved Permanently all future REGISTER messages are sent to the new location Support for multiple Packetization Time Ptime values MP 112 MP 114 MP 118 MP 124 FXS FXO Support for multiple packetizatio
33. FileURL TxDTMFOption 1 to 5 Tx DTMF Option DelayBeforeDIDWink Delay Before DID Wink Send180ForCallWaiting 3xxBehavior NumberOfActiveDialogs Description 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 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 3 INFO Cisco Sends DTMF digits according with Cisco format 4 RFC 2833 5 INFO Nortel Sends DTMF digits according with IETF lt draft choudhuri sip info digit O0 gt 6 NOTIFY Sends DTMF digits according with lt draft mahy sipping signaled digits 01 gt Note 1 DTMF negotiation methods are prioritized according to the order of their appearance Note 2 When out of band DTMF transfer is used 3 5 or 6 the parameter DTMFTransportType is automatically set to 0 DTMF digits are erased from the RTP stream Note 3 When RFC 2833 2 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 re
34. IONS PRACK UPDATE PUBLISH SUBSCRIBE Supported Yes Yes Yes Yes 2 3 3 SIP Headers The following SIP Headers are supported by the MediaPack SIP gateway Header Field Accept Accept Encoding Alert Info Table 2 1 SIP Functions Supported Yes Yes Third party only tested with amongst others Ubiquity Delta3 Microsoft 3Com Snom and Cisco Proxies Third party Third party Yes Third party Table 2 2 SIP Methods Comments Send only Send only Table 2 3 SIP Headers continues on pages 40 to 42 Supported Yes Yes Yes SIP Release Notes 40 Document LTRT 65608 SIP Release Notes 2 SIP Compatibility Table 2 3 SIP Headers continues on pages 40 to 42 Header Field Supported Allow Yes Also Yes Asserted Identity Yes Authorization Yes Call ID Yes Call Info Yes Contact Yes Content Disposition Yes Content Encoding Yes Content Length Yes Content Type Yes Cseq Yes Diversion Yes Encryption No Expires Yes Fax Yes From Yes History Info Yes Join Yes Max Forwards Yes Messages Waiting Yes MIN SE Yes Organization No P Asserted Identity Yes P Preferred Identity Yes Priority No Proxy Authenticate Yes Proxy Authorization Yes Proxy Require Yes Prack Yes Record Route Yes Refer To Yes Referred By Yes Replaces Yes Require Yes Remote Party ID Yes Response Key Yes Retry After Yes Route Yes Rseq Yes Session Expires Yes Server Yes SIP If Match Yes Subject Yes Supported Ye
35. MP 124 FXS The user can now enable the sending of the NTT Japanese standard Direct Inward Dialing DID as a DTMF stream Relevant parameter NTTDIDSignallingForm SIP Release Notes 8 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 6 T 38 packet duplication as No Op packet for NAT MP 112 MP 114 MP 118 MP 124 FXS FXO To enable Network Address Translator NAT port binding for T 38 streams the gateway now supports the sending of T 38 redundant packets to the remote side including during silence periods per user defined interval in seconds Relevant parameters NoOperationSendingMode NoOpIinterval 7 Syslog server port definition support MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now allows the user to define the port of the Syslog server When no port is defined the default port of 514 is used Relevant parameter SyslogServerPort 8 Jitter Buffer defaults modified to 10 msec MP 112 MP 114 MP 118 MP 124 FXS FXO The default values for the Jitter Buffer DJBufMinDelay and DJBufOptFactor ini file parameters were changed to 10 msec Relevant parameters DJBufMinDelay DJBufOptFactor 9 DTMF twist of 3 dB TBR21 standard MP 112 MP 114 MP 118 MP 124 FXS FXO When generating a DTMF signal a difference of 3 dB between high and low frequencies was added In the next applicable version the difference will be reverted back to 0 dB configurable using new configuration parameter 10 HangOver t
36. S resolution of Fully Qualified Domain Names FQDNs received in SDP The authentication process was improved in order to reduce the number of SIP messages transmitted on the network The very first request to the active Proxy is sent without authorization The Proxy sends a 401 407 response with a challenge This response is saved for further uses A new request is resent with the appropriate credentials Following requests to the active Proxy are sent with credentials calculated from the saved challenge If the Proxy doesn t accept the new request and sends another challenge the old challenge is replaced with the new one It is now possible to configure the value assigned to the Subject header If configured the Subject header is added to all outgoing INVITE messages Relevant parameters SIPSubject The DTMF transport mechanism was improved You can now set the preferred DTMF transport methods according to priority In addition the number of configuration parameters was reduced Relevant parameters TxDTMFOption Handling of the OPTIONS method was improved gt Ifan OPTIONS message is received without a user in the Request URI a 200 OK response is sent gt Ifan OPTIONS message is received with a user in the Request URI the gateway tries to find a corresponding user endpoint If no endpoint is found the gateway responds with 404 Not Found If an endpoint is found but this endpoint is currently busy the gateway responds with 48
37. Security Settings 3 Configuration File 4 IPSec IKE tables 5 Software Upgrade Key 6 Internal Firewall 7 Web Access List 8 Web User Accounts NAA Non Authorized Access Attempt to access the Embedded Web Server with a false empty username or password SPC Sensitive Parameters Value Change Changes made to sensitive parameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 4 ActivityListToLog For example ActivityListT oLog pvc afl dr fb swu ard naa spc we Pr SO NS Internal Firewall Parameters AccessList_Source_IP Source IP AccessList_Net_Mask Mask AccessList_Start_Port AccessList_End_Port Local Port Range AccessList_Protocol Protocol AccessList_Packet_Size Packet Size AccessList_Byte_Rate Byte Rate AccessList_Byte_Burst Burst Bytes 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 protocol type isn t TCP or UDP the entire range must be provided The protocol type e g UDP TCP ICMP ESP or Any or the IANA protocol number in
38. Tel 03 2 667 22 Note The Source IP address can include wildcards The x wildcard is used to represent single digits e g 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 Modification asterisk wildcard supported in IP addresses 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 source phone prefix gt can be single number or a range of numbers 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 Note 5 The IP address can include wildcards 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 SIP Release Notes 28 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name PSTNPrefix Description Modificat
39. aded via the ini file the Embedded Web Server or by using the automatic update mechanism Relevant parameters EnableUserlnfoUsage UserlnfoFileName UserlnfoFileURL It is now possible to configure the string that is used in the SIP request header User Agent and SIP response header Server Relevant parameters UserAgentDisplayInfo SIP Release Notes 56 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 10 11 12 13 14 It is now possible to select the SIP method used for session timer updates Two options are available the currently supported re INVITE request and the new request UPDATE The gateway can receive session timer refreshes using both methods Relevant parameters SessionExpiresMethod The Proxy Keep Alive mechanism can now use REGISTER messages instead of OPTIONS messages Any response received from the Proxy either success 200 OK or failure a 4xx response is considered as if the Proxy is communicating Relevant parameters EnableProxyKeepAlive The gateway now supports dynamic Payload Type PT negotiation for the relevant coders For IP to Tel calls if an incoming INVITE includes a coder that uses a dynamic PT the gateway uses the PT defined by the remote side and ignores its configuration For Tel to IP calls the gateway sets the dynamic PT in the SDP according to its configuration but uses the PT defined by the remote side The gateway now supports reception and DN
40. alid IP address range is in dotted notation xxx xxx xxx xxx The default value is 10 4 10 4 Default GW address to use when booting from the flash to non PPPoE Point to PPPoERecoverDfgwA Point Protocol over Ethernet environments ddress The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 10 4 10 1 Subnet Mask to use when booting from the flash to non PPPoE Point to Point PPPoERecoverSubnet Protocol over Ethernet environments Mask The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 255 255 0 0 Server Name for CHAP authentication PPPoEServerName 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 PPPoEStaticIPAddres this IP address is used during the session E The valid IP address range is in dotted notation xxx xxx xxx xxx The default value is 0 0 0 0 User Name for PAP or Host Name for CHAP authentication PPPoEUserName aie The valid range is a string of up to 47 characters The default value is 0 Differential Services Parameters NetworkServiceClassDiffS Sets the DiffServ value for Network service class content erv A Network QoS The valid range is 0 to 56 The default value is 48 PremiumServiceClassMed Sets the DiffServ value for Premium Media service c
41. all uses a separate TCP connection Relevant parameter EnableTCPConnectionReuse 9 Enhanced Proxy Keep Alive mechanism MP 112 MP 114 MP 118 MP 124 FXS FXO The Proxy Keep Alive mechanism was enhanced by using responses to INVITE requests If the active proxy does not respond to INVITE messages sent by the gateway the proxy is marked as offline This capability is in addition to the use of OPTIONS or REGISTER messages Relevant parameter EnableProxyKeepAlive 10 Support for RFC 4244 An Extension to SIP for Request History Information MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports RFC 4244 An Extension to SIP for Request History Information When enabled History Info headers are added and managed according to the history of the call i e redirection requests and or failure responses Relevant parameter EnableHistorylnfo SIP Release Notes 12 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 11 12 13 14 15 Support for handling forking proxy multiple responses MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway can now handle forking proxy multiple responses When passing through a forking proxy the UAC may receive multiple 1xx 2xx responses The multiple responses can be identified by the usage of different remote tags Therefore each response that has a different remote tag than the responses that have already arrived is saved separately Early Media is o
42. ame MultiPtimeFormat Multiple Packetization Time Format SRV2IP Internal SRV Table EnableGRUU Enable GRUU EnableSemiAttendedTran sfer Enable Semi Attended Transfer Description 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 attribute 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 Defines the internal SRV table 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 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 foun
43. ameter Name HookFlashOption Hook flash Option Enable3WayConference Enable 3 Way Conference ConferencelD Conference ID ConferenceCode Establish Conference Code 3WayConferenceMode WarningToneDuration FXONumberOfRings Description Supported hook flash Transport Type method by which hook flash is sent and received Valid options include 0 Hook Flash indication isn t sent default 1 Send proprietary INFO message with Hook Flash indication 4 RFC 2833 Note FXO gateways support the receiving of RFC 2833 Hook Flash signals Enables or disables the 3 Way Conference feature Valid options include 0 Disable default 1 Enable 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 For example ConferencelD MyConference 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 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
44. are sent to the Syslog Values are shown which are offset from the values entered in the Web e g when VoiceVolume is set to X the Syslog message indicates the value X 32 When configuring the RadiusAuthServerlp parameter with a non existent server IP the BehaviorUponRadiusTimeout parameter value is ignored Wrong presentation of dynamic payload in the channel status page The value of Fax Modem Bypass Coder Type in the Web is absent SNMPv3 users table returns line removed notice when adding a new row to an active row index Firefox mozilla Part of the port Info box of Ch 1 4 is obscured by the left menu After adding an empty line to SNMPV3 table it is impossible to delete it or add new lines Wizard gets stuck upon attempt to load an inappropriate file type Firefox Mozilla Port info text box opens too far away from the port Unintended password reset when changing the username and or password via the Web and Reset board again from Web When performing a GET Complete ini file via the Web swwd messages appear 3 5 SNMP Constraints 1 An Ethernet link trap is sent before link is up manager does not receive clear This occurs because a spanning tree algorithm is being calculated in the Ethernet switch The acBoardConfigurationError alarm trap generated as a result of a configuration error does not clear The range of the faxModemRelayVolume MIB object is wrong Instead of O to 15 it should be 18 to 3 corres
45. ases the call sending appropriate release cause to PSTN side The GW generates 5xx response according to PSTN release cause coming from PSTN Version 5 0 45 December 2006 fal AudioCodes MediaPack Series 2 3 5 6 6xx Response Global Responses 600 603 604 606 Table 2 10 6xx SIP Responses 6XX Response Comments Busy Everywhere Decline On reception of any of these Responses the GW releases the call sending appropriate release cause to Does Not Exist Anywhere PSTN side Not Acceptable SIP Release Notes 46 Document LTRT 65608 SIP Release Notes 3 Known Constraints 3 Known Constraints 3 1 Hardware Constraints 1 Only specific combinations of FXS and FXO modules are currently supported For detailed information contact AudioCodes MP 11x After running the procedure for restoring the networking parameters to their initial state the gateway must be reset again using a hardware reset If a software reset is issued the gateway reverts to its factory defaults 3 2 SIP Constraints The Netcoder coder is no longer supported The number of RTP payloads packed in a single G 729 packet M channel parameter is limited to 5 The STUN protocol is used only when the transport protocol is UDP STUN doesn t support TCP In addition STUN doesn t function when the gateway is behind a Symmetric NAT If the gateway is not configured to use RFC 2833 for Hook Flash and the remote side sends su
46. at 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 Version 5 0 59 December 2006 Ta WH gA AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name SessionExpiresMethod Session Expires Method IPProfile_ID IP Profile Settings TelProfile_ID Tel Profile Settings Description Defines the SIP method used for session timer updates 0 Use Re INVITE messages for session timer updates default 1 Use UPDATE messages Note The gateway can receive session timer refreshes using both methods 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 ProgressIndica
47. ateway 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 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 2 3 5 5 5xx Response Server Failure Responses 500 501 502 503 504 505 5xx Response Internal Server Error Not Implemented Bad gateway Service Unavailable Gateway Timeout Version Not Supported Table 2 9 5xx SIP Responses Comments On reception of any of these Responses the GW rele
48. 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 call The valid range is 0 to 255 The default is 0 seconds SIP Release Notes 24 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 1 7 Modified Parameters Table 1 2 lists existing parameters that have been modified Note that only those
49. ceive 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 TXxDTMFOption parameter up to five options can be provided 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 0 Use 182 Queued response to indicate a call waiting default 1 Use 180 Ringing response to indicate a call waiting 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 CalllD 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 Defines the maximum number of active SIP dialogs that are not call related i e REGISTER and SUBSCRIBE This parameter is used to control the Registration Subscription rate The valid range is 1 to 5 The default value is 5 SIP Release Notes 66 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 7
50. ch a signal it is disregarded In addition if the gateway initiates the SIP INVITE with RFC 2833 supported and the remote side does not support this method the gateway still generates the Hook Flash signal 3 3 Gateway Constraints 1 The device attempts to access the incorrect TFTP server when IniFileUrl specifies a TFTP URL It is possible to work around this problem by resetting the device using the Web interface after the TFTP error occurs RFC 2198 redundancy mode with RFC 2833 is not supported that is if a complete DTMF digit was lost it is not reconstructed The current RFC 2833 implementation does support redundancy for inter digit information lost Date and Time should be set after each gateway power reset unless NTP Network Time Protocol is used After resetting the Web password using the ini file parameter ResetWebPassword and defining a new password the user must load an ini file with ResetWebPassword set to 0 Channel parameters such as Voice DTMF gain and Jitter buffer are collectively configured in the ini file on a per gateway usage not on a per call basis By using Profiles this limitation can be overcome The gateway only supports symmetrical coders the same coder is used for transmit and for receive though different ptime is supported Version 5 0 47 December 2006 Ta WH L ol AudioCodes MediaPack Series 10 11 12 13 14 15 16 17 18 19 The following con
51. cribed in Table 4 1 can be configured with the ini file and via the Embedded Web Server Note that only those parameters contained within square brackets are configurable via the Embedded Web Server Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name EnablelPSec Enable IP Security EnableSTUN Enable STUN STUNServerPrimaryIP STUN Server Primary IP STUNServerSecondaryIP STUN Server Secondary IP NATBindingDefaultTimeo ut RTPNoOpEnable RTPNoOpinterval RTPNoOpPayloadType UseSIPTgrp Use Tgrp Information UserAgentDisplaylInfo User Agent Information Description Enables disables the Secure Internet Protocol IPSec on the gateway 0 Disable default 1 Enable 0 STUN protocol is disabled default 1 STUN protocol is enabled When enabled the gateway functions as a STUN client and communicates with a STUN server located in the public 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 port numbers that the NAT assigns to outgoing signaling messages using SIP and media streams using RTP RTCP and T 38 STUN works with many existing NAT types and does not require any special behavior from them This parameter cannot be changed on the fly and requires a gateway reset The IP address of the primary STUN
52. d the gateway performs an SRV resolution using an external DNS server Note 2 This parameter can appear up to 10 times Determines whether or not the Globally Routable User Agent GRUU mechanism is used Valid options include 0 Disable default 1 Enable The gateway obtains a GRUU by generating a normal REGISTER request If the Registrar Proxy supports GRUU the REGISTER responses contain the gruu parameter in each Contact header field The gateway includes the GRUU in any header field that 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 the Address of Record 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 Determines the gateway s behavior when Transfer is initiated while still in Alerting state Valid options include 0 Send REFER with Replaces default 1 Send CANCEL and after a 487 response is received send REFER without Replaces Version 5 0 23 December 2006 Ta WH L al AudioCodes MediaPack Series Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Par
53. d 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 Version 5 0 21 December 2006 Ta WH gA AudioCodes MediaPack Series Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Parameter Name EnableHistorylinfo Enable History Info Header HookFlashCode Hook Flash Code EnableTCPConnectionRe use Enable TCP Connection Reuse 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 Q 850 Reason SIP Reason SIP Response code Upon receiving the final success or failure response the gateway searches for a Redirect reason in the Histo
54. d or long hex string Keys are always persisted as long hex strings and keys are localized Privacy key Keys can be entered in the form of a text password or long hex string Keys are always persisted as long hex strings and keys are localized The group with which the SNMP v3 user is associated Valid options include 0 Read only group default 1 Read write group 2 Trap group Note All groups can be used to send traps 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 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 19 December 2006 Ta WH L al AudioCodes MediaPack Series Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Parameter Name EnableReasonHeader Enable Reason Header ProxyDNSQueryType Proxy DNS Query Type DNSQueryType DNS Query Type Description Enables or disables the usage of the SIP Reason header Valid options include 0 Disable 1 Enable default
55. ed and P Preferred headers 0 sip default 1 tel Applicable to Tel gt IP calls 0 No 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 1 Yes If a Tel Display Name is received the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name If no Display Name is received from the Tel side the Tel Source Number is used as the IP Source Number and also as the IP Display Name 2 Overwrite The Tel Source Number is used as the IP Source Number and also as the IP Display Name even if the received Tel Display Name is not empty Applicable to P gt Tel calls 0 No 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 1 Yes 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 S
56. eived 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 receiving this message the GW issues a new request according to the scheme received on this response The gateway generates this response if no answer timeout expired 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 before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On recep
57. election algorithm in addition to the existing By Destination Phone Number option Relevant parameters ChannelSelectMode TrunkGroupSettings Support for sending CDR on call connect MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway can now send a CDR when a call is connected in addition to when the call ends Relevant parameter CDRReportLevel Supports Call Forward Indication to user MP 112 MP 114 MP 118 MP 124 FXS O FXO When a phone on an FXS port goes off hook to make a call the gateway typically plays a regular dial tone for a period of time indicating that the user can dial digits If this port has Call Forwarding active the user needs to be aware that any calls made to their directory number DN are not sent to their phone To notify users of the call forwarding status this Call Forward Indication feature plays a stutter dial tone instead of the regular dial tone This indicates to the user that to receive calls the user needs to disable call forwarding Relevant parameter Stutter ToneDuration Supports defining of Off Hook Warning Tone duration MP 112 MP 114 MP 118 MP 124 FXS FXO It is now possible to set the duration for which the Off Hook Warning Tone is played Relevant parameter WarningToneDuration Support for defining number of rings before FXO answers MP2 MPAIAMP 118 MPAA O PXS FXO It is now possible to define the number of rings before the FXO gateway answers a call Relevant parameter
58. es into a conference session The first Hook Flash action places the first initial call on hold and then answers the second call Another Hook Flash sends a Conference initiating INVITE to a media server At the same time the gateway sends REFER messages to the two remote parties to add them as participants to the conference It s possible to configure the digit pattern that once detected generates the Conference initiating INVITE by default it s set to Hook Flash In addition there are two different modes of operation gt The Conference initiating gateway provides the conference unique identifier gt The media server provides the conference unique identifier and the gateway distributes this identifier to the remote parties in the Refer To header of the REFER message Relevant parameters Enable3WayConference ConferencelD ConferenceCode 3WayConferenceMode Call Waiting for Remote Extension MP 112 MP 114 MP 118 MP 124 FXS FXO When the FXO gateway detects a Call Waiting indication FSK data of the Caller Id it sends a proprietary INFO message which includes the caller identification to the FXS gateway Once the FXS gateway receives this INFO message it forwards the information to the relevant port for display 1 4 Web and SNMP New Features 1 Supports Graceful Forced Lock and Reset MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports a lock unlock mechanism The user can select one of the follow
59. estination 305 Use Proxy Yes The gateway responds with an ACK and resends the request to new address 380 Alternate Yes Service Version 5 0 43 December 2006 Ca AudioCodes MediaPack Series 2 3 5 4 4xx Response Client Failure Responses 400 401 402 403 404 405 406 407 408 409 410 411 413 414 415 420 Table 2 8 4xx SIP Responses continues on pages 44 to 45 4xx Response Bad Request Unauthorized Payment Required Forbidden Not Found Method Not Allowed Not Acceptable Proxy Authentication Required Request Timeout Conflict Gone Length Required Request Entity Too Large Request URL Too Long Unsupported Media Bad Extension 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 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 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 rec
60. fferent IP and Tel Profiles was _ increased to 10 each Relevant parameters IPProfile TelProfile 5 FXS gateways now support generation of 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 includes the time interval between the generated pulses and the number of pulses generated on answer Relevant parameters ChargeCode MeteringType PayPhoneMeteringMode Prefix 6 The RADIUS Accounting mechanism is now supported The gateway sends a CDR to the RADIUS Accounting Server at the start and or end of each call Relevant parameters AAAIndications RADIUSAccServerlP RADIUSAccPort RADIUSAccountingType 7 FXO gateways can now disconnect a call after a dial tone from the PBX is detected This is in addition to the existing capability of call disconnection when either busy or reorder tones are detected Relevant parameters DisconnectOnDialTone SIP Release Notes 54 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 10 11 12 13 14 15 16 17 18 19 20 FXO gateways now support a guard time between accepting successive IP to Tel calls Occasionally after a call is ended and onhook is applied a delay is required before placing a new cal
61. he 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 6 By Source Phone Number Select the gateway port according to the calling number The default method is By Phone Number SIP Release Notes 30 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name Description TxDTMFOption Modification enumeration order and default value istto 5th Tx DTMF Option Determines a single or several preferred transmit DTMF negotiation methods Valid options include 0 Not Supported No negotiation DTMF digits are sent according to the parameters DTMFTransportType and RFC2833PayloadT ype 1 INFO Nortel Sends DTMF digits according to IETF lt draft choudhuri sip info digit 00 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 default Note 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 DTMFTransportType is automatically set to 0 DTMF digits are erased from the RTP stream
62. he 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 SIP Release Notes 34 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 1 8 Obsolete Parameters Table 1 3 lists parameters from the previous release that are no longer in use Table 1 3 Release 5 0 Obsolete ini File Web Parameters ini File Web Interface Parameter Name EnableProxySRVQuery EnableSRVQuery RTPNoOpEnable RTPNoOpinterval IsHookFlashUsed Description This parameter is obsolete use instead the parameter ProxyDNSQueryT ype This parameter is obsolete use instead the parameter DNSQueryType This parameter is obsolete use instead the parameter NoOperationSendingMode This parameter is obsolete use instead the parameter NoOpInterval This parameter is obsolete use instead the parameter HookFlashOption Version 5 0 35 December 2006 MediaPac Reader s Notes SIP Release Notes 36
63. ication Support for additional coders T 38 G 711A law_VBD and G 711U law_VBD G 726 payload type and rate modified Defines the gateway s coder list up to five coders can be configured 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 Coder Name Packetization Rate Payload Type Silence Time Suppression G 711 A law 10 20 default Always 64 Always 8 Disable 0 g711Alaw64k 30 40 50 60 Enable 1 80 100 120 G 711 p law 10 20 default Always 64 Always 0 Disable 0 g711Ulaw64k 30 40 50 60 Enable 1 80 100 120 G 729 10 20 default Always 8 Always 18 Disable 0 9729 30 40 50 60 Enable 1 80 100 Enable w o Adaptations 2 Version 5 0 25 December 2006 Ta WH gA AudioCodes MediaPack Series Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Description Parameter Name p G 723 1 30 default 60 5 3 0 Always 4 Disable 0 97231 90 120 6 3 1 default Enable 1 G 726 10 20 default 16 0 24 1 Dynamic 0 Disable 0 9726 30 40 50 60 32 2 default 120 Enable 1 80 100 120 40 3 T 38 N A N A N A N A t38fax G 711A 10 20 default Always 64 Dynamic 0 N A law_VBD 30 40 50 60 120 g711AlawVbd 80 100 120 G 711U 10 20 default Always 64 Dynamic 0 N A law_VBD 30 40 50 60 1
64. ication additional enumeration value for By Source Phone Number for selecting port Defines common rule for port allocation of IP to Tel calls Valid options include 0 By phone number Select the gateway port according to the called number called number is defined in the Endpoint Phone Number table 1 Cyclic Ascending 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 2 Ascending 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 3 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 4 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 5 Number Cyclic Ascending First select the gateway port according to the called number called number is defined in the Endpoint Phone Number table If t
65. igured in the Web server The search can be performed for a specific ini parameter e g EnablelPSec or a sub string of the 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 SNMPv3 support MP 112 MP 114 MP 118 MP 124 FXS FXO In previous releases it was assumed that customers could use SNMPv2c over IPsec to meet their SNMP security requirements While SNMP over IPsec is a viable solution for some customers others however demand SNMPv3 security Therefore in Release 5 0 support for SNMPv3 authentication and privacy has been provided This feature allows customers to 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 The customer still has the option for defining up to five read only community strings and up to five read write community strings Relevant parameters SNMPUsers_Index SNMPUsers_ Username SNMPUsers_AuthProtocol SNMPUsers_PrivProtocol SNMPUsers_AuthKey SNMPUsers_PrivKey SNMPUsers_Group Resolved Constraints N A Version 5 0 17 December 2006 7a Va a gA AudioCodes MediaPack Series 1 6 New Parameters Most new parameters described in Table 1 1 can be configured with the ini file and via the Embedded Web Serve
66. ime setting after muting DTMF or MF from Tel PSTN MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now allows the user to define the HangOver time which is the voice silence time in msec after muting DTMF or MF digits received from the Tel PSTN side before sending to the IP side Relevant parameter TxDTMFHangOverTime 11 HangOver time setting after playing DTMF or MF from IP network MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now allows the user to define the HangOver time which is the voice silence time in msec after playing DTMF or MF digits received as Relay from the IP side to the Tel PSTN side Relevant parameter RxDTMFHangOverTime 12 IP addresses support the asterisk wildcard MP 112 MP 114 MP 118 MP 124 FXS FXO The asterisk wildcard can now be included in IP addresses IP addresses that are defined in the Routing tables and Manipulation tables can include the wildcard which represents any valid value from 0 to 255 For example 10 8 8 represents IP addresses from 10 8 8 0 to 10 8 8 255 10 8 represents IP addresses from 10 8 0 0 to 10 8 255 255 Relevant parameters Prefix PSTNPrefix NumberMapIP2Tel SourceNumberMapIP2Tel Version 5 0 9 December 2006 a WH gA AudioCodes MediaPack Series 13 14 15 16 17 Additional option for Channel Selection algorithm MP 112 MP 114 MP 118 MP 124 FXS FXO An additional option By Source Phone Number was added to the channel s
67. ing options gt Graceful Lock the gateway rejects all incoming calls All existing calls are allowed to continue until a user defined lock timer expires When this timer expires the gateway disconnects the calls At this point the gateway remains in an idle state gt Forced Lock the gateway rejects all incoming calls and all existing calls are disconnected At this point the gateway remains in an idle state gt Unlock the gateway returns from Lock state to normal state and accepts incoming calls from Tel and IP sides gt Graceful Reset same behavior as Graceful Lock only that when all calls are disconnected the gateway performs reset gt Forced Reset same behavior as Forced Lock only that when all calls are disconnected the gateway performs reset SIP Release Notes 16 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 2 1 5 Support for assigning free text description for each port MP 112 MP 114 MP 118 MP 124 FXS FXO This feature allows users to assign free text descriptions for each port via the gateway s Embedded Web Server Web Search Engine support MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway s Embedded Web Server now provides a search engine Search button for searching any ini file parameter that is configurable by the Web server The search result provides you a brief description of the parameter as well as a link to the relevant screen in which the parameter is conf
68. ion asterisk wildcard supported in IP addresses The format for PSTNPrefix is as follows a b c d e Where a 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 5 This parameter can appear up to 24 times Version 5 0 29 December 2006 Ta Ca AudioCodes MediaPack Series Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name ChannelSelectMode Channel Select Mode Description Modif
69. l and performing offhook This is necessary to prevent wrong hook flash detection or other glare phenomena Relevant parameters GuardTimeBetweenCalls FXS gateways can now add a delay between detection of offhook and generation of DID Wink Relevant parameters DelayBeforeDIDWink It is now possible to determine 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 Up to this version the gateway played a reorder tone to the connected phone PBX It is now possible to set the behavior of such endpoints to either no response reorder tone polarity reversal or both Relevant parameters FXSOOSBehavior It is now possible to define 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 Relevant parameters TelConnectCode The parameter Source Number before Manipulation was added to CDR messages The MediaPack can now 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 Relevant Parameters NetworkServiceClassDiffServ PremiumServiceClassMediaDiffServ PremiumServiceClassControlDiffServ GoldServiceClassDiffServ BronzeServiceClassDiffServ
70. lass content only if IPDiffServ iaDiffServ is not set in the selected IP Profile Media Premium QoS 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 PremiumServiceClassCon_ Sets the DiffServ value for Premium Control service class content only if trolDiffServ ControllPDiffserv is not set in the selected IP Profile Control Premium QoS 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 Version 5 0 69 December 2006 E a C wt AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Description Parameter Name p GoldServiceClassDiffServ Sets the DiffServ value for the Gold service class content The valid range is 0 to 56 The default value is 26 Gold QoS BronzeServiceClassDiffSe Sets the DiffServ value for the Bronze service class content y The valid range is 0 to 56 The default value is 10 Bronze QoS 4 7 Version History Details of previous releases can be found in the Release Notes of Version 4 6 published by AudioCodes on Jul 13 2005 SIP Release
71. lnet sessions are now supported Version 5 0 55 December 2006 a WH gA AudioCodes MediaPack Series 21 22 23 24 25 Support for DTMF relay according to RFC 2833 was added to the ThroughPacket mechanism An Activity Log mechanism was added to enable 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 Auxiliary Files Loading Device Reset Flash Memory Burning Device Software Update Access to Restricted Domains Non Authorized Access and Sensitive Parameters Value Change Relevant parameter ActivityListToLog The automatic update mechanism enables loading files also via FTP FTPS and Network File System NFS Initial configuration of the gateway can now 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 Relevant parameter VoiceMenuPassword As of this version it isn t required to load a coefficients file to FXO gateways instead there is a single parameter that defines the country variant doesn t apply to the Mediant 1000 Relevant parameter CountryCoefficients 4 3 SIP New Features 1 MediaPack only The gateway now supports Secured RTP SRTP according to RFC 3711 SRTP is used to encrypt RTP and RTCP transport SRTP requi
72. n time Ptime values was added RFC 2327 SDP only defines a ptime attribute that includes the value for the first defined coder per media line A new SDP attribute called mptime has been added in which a separate ptime value is defined for each coder provided in the media line Relevant parameter MultiPtimeFormat Enhanced support for G 726 32 kbps coder MP 112 MP 114 MP 118 MP 124 FXS FXO For coder G 726 32 kbps the gateway now supports a dynamic payload type instead of the constant one 2 In addition support was added for rates of 16 24 and 40 kbps Relevant parameters CoderName CoderName_ID Support for RFC 3389 RTP Payload for Comfort Noise MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports RFC 3389 RTP Payload for Comfort Noise When using the SDP to specify RTP payload information the use of comfort noise CN is indicated by including a payload type for CN on the media description line The gateway can use CN with a codec whose RTP timestamp clock rate is 8 000 Hz G 711 G 726 The use of CN is negotiated between sides therefore if the remote side does not support CN it s not used Relevant parameter ComfortNoiseNegotiation Version 5 0 11 December 2006 Ta Db wt AudioCodes MediaPack Series 6 Support for RFC 3959 The Early Disposition Type for SIP MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway supports RFC 3959 The Early Disposition Type for SIP The gateway
73. nerates 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 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 Prefix lt Destination Phone Prefix gt lt IP Address gt lt Src Phone Prefix gt lt IP Profile ID gt lt Charge Code gt Selection of IP address for Tel To IP calls is according to destination and source prefixes Note 1 An optional IP ProfilelD 1 to 9 can be applied to each routing rule Note 2 An optional Charge Code 1 to 25 can be applied to each routing rule to associate it with an entry in the ChargeCode table SIP Release Notes 62 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name EnableMediaSecurity Enable Media Security MediaSecurityBehaviour Media Security Behavior Description Enables or disables the Secure Real Time Transport Protocol SRTP 0 SRTP is disabled default 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 SRTP isn
74. ng AAA indications that are used 0 No indications default 3 Accounting only IP address of accounting server Port number of RADIUS accounting server The default value is 1646 Determines when a RADIUS accounting report is issued 0 At the release of the call only default 1 At the connect and release of the call 2 At the setup and release of the call Defines the value of the Subject header in outgoing INVITE messages If not specified the Subject header isn t included default The maximum length of the subject is limited to 50 characters 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 Enables or disables usage of the User Information loaded to the gateway via the User Information auxiliary file 0 Disable default 1 Enable The name and path of the file containing the User Information data Version 5 0 65 December 2006 Ta WH gA AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name UserlInfo
75. nly established with the first received 1xx response if necessary The first transaction that receives a 200 OK response is the active one For other 200 OK responses the gateway sends an ACK and immediately afterwards sends a BYE DNS Naming Authority Pointer NAPTR queries for resolving domain names MP 112 MP 114 MP 118 MP 124 FXS FXO DNS Naming Authority Pointer NAPTR queries can now be used to resolve domain names of Proxy servers Registrar servers and any domain name that appears in the Contact and Record Route headers Relevant parameters DNSQueryType ProxyDNSQueryType Supports Load Balancing for Proxy servers MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway can use load balancing algorithms when using Proxy servers The load balancing can either use a Round Robin mechanism where each new request is sent to the next Proxy in the list or use a Random Weights algorithm according to the SRV response received from the domain name system DNS server When weights are used the requests are not distributed equally among the servers but according to the weight assigned to each server Relevant parameters ProxyLoadBalancingMethod ProxylPListRefreshTime Supports NAPTR SRV resolution on Routing table entries MP 112 MP 114 MP 118 MP 124 FXS FXO If the destination IP address in the Tel2IP Routing table or the source IP address in the IP2Tel Routing table is a fully qualified domain name FQDN and the gateway is configured to
76. now supports two separate Offer Answer sessions one for regular voice establishment the other for an Early Media session The gateway doesn t generate two SDP Offers but it supports reception of two Offers and responds appropriately To support this function both Early Media and PRACK usage must be configured Relevant parameters EnableEarlyMedia PRACKMode 7 Support for ITU V 152 Procedures for supporting Voice Band Data over IP Networks MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports ITU V 152 Procedures for supporting Voice Band Data over IP Networks Voice Band Data VBD is the transport of modem fax and text telephony signals over a voice channel in a packet network with a codec appropriate for such signals The coder table configuration was enhanced to support the V 152 capability New entries were added to indicate T 38 and VBD support either A law or Mu law All selected options are declared in the outgoing SDP when initiating calls to IP Negotiation is performed separately on each capability voice coder VBD and T 38 After the initial negotiation no Re INVITEs are necessary as both sides are synchronized in terms of the other side s capabilities Relevant parameters CoderName CoderName_ID IsFaxUsed 8 Support for TCP connection re use MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports the reuse of the same TCP connection for all calls to the same destination When disabled each c
77. nt 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 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 28 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 Defines a digit pattern that is sent to the Tel side after 200 OK is received from the IP side
78. onses Table 2 5 1xx SIP Responses Comments The SIP gateway generates this response immediately after receiving an INVITE request The SIP gateway generates this response for an incoming INVITE message On receiving this response the gateway waits for a 200 OK response The SIP gateway does not generate these responses However the gateway does receive them The gateway processes these responses the same way that it processes the 100 Trying response The SIP gateway generates this response in Call Waiting service When SIP gateway receives 182 response it plays a special waiting Ringback tone to Tel side The SIP gateway generates this response if Early Media feature is enabled and if the gateway plays a Ringback tone to IP 2 3 5 2 2xx Response Successful Responses 200 202 2xx Response OK Accepted Table 2 6 2xx SIP Responses Supported Comments 2 3 5 3 3xx Response Redirection Responses Table 2 7 3xx SIP Responses 3xx Response Supported Comments 300 Multiple Choice Yes The gateway responds with an ACK and resends the request to first in the contact list new address 301 Moved Yes The gateway responds with an ACK and resends the request to Permanently new address 302 Moved Yes The SIP gateway generates this response when call forward is Temporarily used to redirect the call to another destination If such response is received the calling gateway initiates an INVITE message to the new d
79. or T 38 No Op packets are sent in the case of silence no RTP T 38 traffic wnen 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 SyslogServerPort Defines the UDP port of the Syslog server Syslog Server Port The valid range is 0 to 65 535 The default port value is 514 PPPoELCPEchoEnable 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 RxDTMFHangOverTime 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 Release Notes 18 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Paramete
80. ource 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 Number is set to 100 and the Presentation is set to Restricted 1 Version 5 0 61 December 2006 Ta WH gA AudioCodes MediaPack Series Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name MeteringType Metering Tones Type PayPhoneMeteringMode Generate Metering Tones ChargeCode Charge Codes Table Prefix Tel to IP Routing Table Description 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 Determines the method used to configure the metering tones that are generated to the Tel side FXS gateways only 0 disabled Metering tones aren t generated default 1 internal table Metering tones are generated according to the internal table configured by the parameter ChargeCode 2 RADIUS N A Note This parameter is not applicable to the Metering Tones Relay mechanism The charge code table is used to configure the metering tones and their time interval that the FXS gateway generates to the Tel side Up to 25 different
81. parameters contained within square brackets are configurable via the Embedded Web Server Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name DJBufMinDelay Dynamic Jitter Buffer Minimum Delay DJBufOpitFactor Dynamic Jitter Buffer Optimization Factor IKEPolicyProposalEncryption_ X First to Fourth Proposal Encryption Type IPSecPolicyProposalEncryption X First to Fourth Proposal Encryption Type Description Modification default value Minimum delay for the Dynamic Jitter Buffer The valid range is 0 to 150 msec default is 10 Modification default value Dynamic Jitter Buffer frame error delay optimization factor The valid range is 0 to 13 default is 10 Note Set to 13 for data fax and modem calls Modification additional enumeration value for AES support Determines the encryption type used in the main mode negotiation for up to four proposals X denotes the proposal number 0 to 3 Valid options include Not Defined default 1 DES CBC 2 Triple DES CBC 3 AES Modification additional enumeration value for AES support Determines the encryption type used in the quick mode negotiation for up to four proposals X denotes the proposal number 0 to 3 Valid options include Not Defined default Q None no encryption 1 DES CBC 2 Triple DES CBC 3 AES CoderName Modif
82. ponding to an actual volume of 18 5 dBm to 3 5 dBm Cold start trap doesn t appear after soft reset for MediaPack Only one SNMP manager can access the device simultaneously SIP Release Notes 50 Document LTRT 65608 SIP Release Notes 3 Known Constraints The default values created in an IPSec configuration table are wrong The user should override the default values before activating the new row Only one SNMP manager can access the boards modules at one time The following RTP MIB objects are not supported gt rtpRcvrSRCSSRC gt rtpRevrSSRC gt rtpSenderSSRC gt rtpRevrLostPackets gt rtpRcevrPackets gt rtpSenderPackets gt rtpRcvrOctets gt rtpSenderOctets The following encryptions types are currently supported for SNMP v3 users only gt Authentication protocol MD5 and SHA gt Privacy protocol DES and AES128 Version 5 0 51 December 2006 MediaPac Reader s Notes SIP Release Notes 52 Document E LTRT 65608 SIP Release Notes 4 Previous Release 4 8 4 Previous Release 4 8 4 1 Supported Hardware Platforms 4 1 1 New Hardware Platforms Introduced in This Release The following hardware platforms are introduced in this version MP 118 FXO with 8 FXO ports and MP 114 FXO with 4 FXO ports MP 118 FXS FXO with 4 FXS ports and 4 FXO ports This gateway contains a relay that connects the FXS ports to the FXO ports in case of a power failu
83. r between 0 and 255 e g 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 Version 5 0 27 December 2006 Ta Ca AudioCodes MediaPack Series Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name SourceNumberMapIP2Tel Prefix Description Modification asterisk wildcard supported in IP addresses Manipulates the source number for IP to Tel calls The format for SourceNumberMapIP2Tel is as follows a b c d e f g h Where 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 a f Notin use should be set to g Destination number prefix h Notin use should be set to 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 T conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example SourceNumberMapIP2Tel 01 2 972 034 SourceNumberMapIP2
84. r Note that only those parameters contained within square brackets are configurable via the Embedded Web Server Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Parameter Name Description MaxEchoCancellerLength Maximum Echo Canceller length in msec Max Echo Canceller Length valid options include 0 based on various internal gateway settings 64 msec 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 NTTDIDSignallingForm Determines the type of Direct Inward Dialing DID signaling support for NTT Japan 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 Valid options include 0 FSK based signaling default 1 DTMF based signaling Note Applicable only to FXS gateways NoOperationSendingMod Enables or disables the transmission of RTP or T 38 No Op packets e Valid options include 0 Disable default 1 Enable This mechanism ensures that the NAT binding remains open during RTP or T 38 silence periods NoOplinterval Defines the time interval in which RTP
85. r Name TxDTMFHangOverTime STUNServerDomainName SNMP V3 Settings SNMPUsers_Index Index SNMPUsers_Username Username SNMPUsers_AuthProtocol AuthProtocol SNMPuUsers_PrivProtocol PrivProtocol SNMPUsers_AuthKey AuthKey SNMPUsers_PrivKey PrivKey SNMPUsers_Group Group SIP Parameters ComfortNoiseNegotiation Comfort Noise Generation Negotiation Description 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 is either Relay or Mute The Valid range is 0 to 2 000 msec The default is 100 msec Defines the domain name for the STUN server s address 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 Use either the STUNServerPrimarylIP or STUNServerDomainName method with priority to the former one SNMP v3 user table index The valid range is 0 to 9 Name of the SNMP v3 user This name must be unique Authentication protocol for the SNMP v3 user Valid options include Q None default 1 MD5 2 SHA 1 Privacy protocol for the SNMP v3 user Valid options include Q None default 1 DES 2 3DES 3 AES128 4 AES192 5 AES256 Authentication key Keys can be entered in the form of a text passwor
86. re MediaPack MP 124 FXS Rev D 24 analog FXS interfaces 4 1 2 Existing Hardware Platforms Analog Mediant 1000 hosting FXS or FXO modules up to 4 ports in each module with a total of 6 modules providing up to 24 ports MediaPack MP 11x FXS 2 to 8 analog FXS interfaces with enhanced CPU resources gt MediaPack MP 118 FXS 8 analog FXS interfaces gt MediaPack MP 114 FXS 4 analog FXS interfaces gt MediaPack MP 112 FXS 2 analog FXS interfaces 4 1 3 Hardware Platforms No Longer Supported MediaPack MP 108 FXS 8 analog FXS interfaces MediaPack MP 108 FXO 8 analog FXO interfaces MediaPack MP 104 FXS 4 analog FXS interfaces MediaPack MP 104 FXO 4 analog FXO interfaces MediaPack MP 102 FXS 2 analog FXS interfaces MediaPack MP 124 FXS Rev A B C 24 analog FXS interfaces Version 5 0 53 December 2006 7a Va a gA AudioCodes MediaPack Series 4 2 General Gateway New Features 1 Support for the IPSec and IKE protocols was added IPSec and IKE are part of the IETF standards for establishing a secured IP connection between two applications Providing security services at the IP layer IPSec and IKE are transparent to IP applications IPSec and IKE are used in conjunction to provide security for control e g SIP and management e g SNMP and Web protocols but not for media i e RTP RTCP and T 38 Relevant Parameters EnablelPSec and the following table parameters IPSEC_IKEDB_TABLE IPSEC_SPD_TABLE for
87. res 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 header 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 of SRTP may reduce the number of available channels Relevant parameters EnableMediaSecurity MediaSecurityBehaviour Support was added for hunt group usage and declaration according to lt draft ietf iptel trunk group 04 gt If enabled the hunt group number is added as the tgrp parameter to the Contact header of outgoing SIP messages For incoming SIP messages if the Request URI includes a tgrp parameter the gateway routes the call according to that value if possible In addition if the incoming Contact header contains a tgrp parameter it is copied to the corresponding outgoing messages in that dialog Relevant parameters UseSIPTgrp A new auxiliary file User Information was added This file contains a list of PBX extensions and their attributes i e global phone number display name user name and password This information is loaded to the gateway and is used to emulate a large number of SIP user agents Each entry defined by the User Information file is registered separately emulating the behavior of an IP phone This file can be lo
88. rm wd AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x Release Notes Version 5 0 Document LTRT 65608 December 2006 SIP Release Notes Contents Table of Contents 1 What s New in Release 5 0 1 1 Supported H 1 1 1 Newt 12 General Gateway Mew Features ccs cicesiccsccesesccssscsesscssscsececcazosaceeseisesatiawciscicaerssivens 1 3 SIP New Features 1 4 Web and SNMP New 1 5 Resolved Constraint isi 10 Nos ii seth needs se betes eee A L7 Modified Paramete Seroicsiconsoniansnnseneni smie R 1 8 Obsolete Parameters c cccccccccsecccseceessececceeecesssecesseecsesseecsstecessstecsteeeesssteessees OO 2 Sl P Co mpati bil ity SERRE RRR RRR RRR of 2 1 Supported SIP Features 2 2 Unsupported SIP Features EEEN P E EN 23 SI P Compl lanes TADES a a a AO 4 2 General Gateway 4 3 SIP New Features 4 4 Web and SNMP N 4 5 Resolved Constraints c eeseeeeeeeee 4 6 Maw an kran arama ERS Version 5 0 3 December 2006 7a T wt AudioCodes __MediaPack Series List of Tables Tiie 1 1 Release a 0 e ini File Web airida Paramena ues on 1 pag S 18 to 24 18 XX SIP oe xx SIP Responses SIP Release Notes 4 Document LTRT 65608 SIP Release Notes Notices Notices Notice This document describes the release of the AudioCodes MediaPack Analog Media Gateway Series MP 124 24 port
89. ry 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 Determines a digit pattern which when received from the Tel side indicates a Hook Flash event The valid range is a 25 character string Enables the reuse of the same TCP connection for all calls to the same destination Valid options include Q Use a separate TCP connection for each call default 1 Use the same TCP connection for all calls SIP Release Notes 22 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Parameter N
90. s Version 5 0 41 December 2006 Ta Db wt AudioCodes MediaPack Series Table 2 3 SIP Headers continues on pages 40 to 42 Header Field Supported Timestamp Yes To Yes Unsupported Yes User Agent Yes Via Yes Voicemail Yes Warning Yes WWW Authenticate Yes 2 3 4 SDP Headers The following SDP Headers are supported by the MediaPack SIP gateway Table 2 4 SDP Headers SDP Header Element Supported 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 2 3 5 SIP Responses The following SIP responses are supported by the MediaPack SIP 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 SIP Release Notes 42 Document LTRT 65608 SIP Release Notes 2 SIP Compatibility 2 3 5 1 1xx Response 100 180 181 182 183 Trying Ringing Call is being forwarded Queued Session Progress Supported Yes Yes Yes Yes Yes 1xx Response Information Resp
91. s that appear in the Contact and Record Route headers Valid options include 0 A Record default 1 SRV 2 NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy Registrar IP address parameter 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 2 an NAPTR query is performed If it is successful an SRV query is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy Registrar IP address parameter 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 SIP Release Notes 20 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 1 Release 5 0 New ini File Web Browser Parameters continues on pages 18 to 24 ini File Web Interface Parameter Name ProxyLoadBalancingMeth od Proxy Load Balancing Method ProxylIPListRefreshTime Proxy IP List Refresh Time Description Enables
92. straints apply when defining coders via the ini file gt Coder names are case sensitive gt Don t use obsolete coder names e g g729_AnnexB g7231r53 with the improved coder interface gt When an invalid packetization time is used the coder definition is disregarded gt When an invalid rate is used for dynamic rate coders the coder definition is disregarded The RFC2833RxPayloadType and RFC2833TxPayloadType parameters in the Embedded Web Servers Channel Settings screen or in the ini file should not be used Use the parameter Rfc2833PayloadType instead Configuring the board to auto negotiate mode while the opposite port is set manually to full duplex either 10 Base T or 100 Base TX is invalid It is also invalid to set the board to one of the manual modes while the opposite port is configured differently It is recommended to use full duplex connections instead of half duplex and 100 Base TX instead of 10 Base T due to the larger bandwidth It is strongly recommended to use 100 Base T switches Use of 10 Base T LAN hubs should be avoided In some cases when the spanning tree algorithm is enabled on the external Ethernet switch port connected to the gateway the external switch blocks traffic entering and exiting the gateway for some time after the gateway is reset This may cause the loss of important packets such as BootP and TFTP requests which in turn may cause the board to fail to start up
93. t ProxyIP 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 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 addresses of the third redundant Proxy is defined by the fourth repetition of the ini file parameter ProxylP RegistrarlP Modification EnableProxySRVQuery changed to ProxyDNSQueryType Registrar IP Address Added NAPTR support IP address and optionally port number of Registrar server 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 ProxyDNSQueryType is set to 1 or 2 Version 5 0 33 December 2006 Ta Ca AudioCodes MediaPack Series Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name TrunkGroupSettings Hunt Group Settings CDRReportLevel CDR Report Level StutterToneDuration Stutter Tone Duration
94. ter 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 Note 4 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 Note 5 If the coder G 729 is selected and silence suppression is enabled for this coder the gateway includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is set to Enable w o Adaptations annexb yes is included An exception to this logic is when the remote gateway is a Cisco device IsCiscoSCEMode Note 6 This parameter CoderName_ID can appear up to 20 times five coders in four coder groups For example CoderName_1 g 711Alaw64k 20 0 CoderName_1 g 711Ulaw64k 40 CoderName_1 g7231 90 1 1 CoderName_2 g726 2 0 FXO gateways can disconnect a call after a dial tone from the PBX is detected 0 Call isn t released 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 Determines the Authentication Authorization and Accounti
95. tform is introduced in this version e MP 114 FXS FXO with 2 FXS ports and 2 FXO ports This product also contains a relay that connects the FXS ports to the FXO ports in case of a power failure 1 1 2 Support for Existing Hardware Platforms e MediaPack MP 118 FXS FXO with 4 FXS ports and 4 FXO ports e MediaPack MP 11x FXS 2 to 8 analog FXS interfaces with enhanced CPU resources gt MediaPack MP 118 FXS 8 analog FXS interfaces gt MediaPack MP 114 FXS 4 analog FXS interfaces gt MediaPack MP 112 FXS 2 analog FXS interfaces e MediaPack MP 124 FXS 24 analog FXS interfaces 1 1 3 Hardware Platforms No Longer Supported N A Version 5 0 T7 December 2006 7a Va a wt AudioCodes MediaPack Series 1 2 General Gateway New Features 1 STUN server support for DNS SRV MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now allows the user to provide a domain name for the STUN server s address The STUN client can perform the required SRV query to resolve the domain name to an IP address and port sort the server list and use the servers according to the sorted list Relevant parameter STUNServerDomainName 2 Separate interfaces for management and conirol bearer MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway can now be configured to operate in Dual IP mode by assigning the same IP address to two traffic types When operating with multiple IP interfaces the gateway splits the traffic into three types Management OAM
96. the Channel Status screen The Fax Modem Bypass Packing Factor field doesn t support the G726_32 and G726_40 options After resetting the Web password using the ini file parameter ResetWebPassword and defining a new password the user must load an ini file with ResetWebPassword set to 0 The Embedded Web Server cannot be accessed with HTTPS when DES 3DES is selected on Microsoft Internet Explorer If the gateway is configured to use the DES cipher a logic error in Microsoft Internet Explorer causes the HTTPS connection to fail This problem does not occur when using alternative browsers such as Firefox Username and password with 8 characters cannot be entered Not all parameters can be changed on the fly from the Web browser Parameters that can t be changed on the fly are noted with To change these parameters reset the gateway using the Web browser reset button When changing gateway parameters from the Web browser the new parameters are permanently stored in flash memory only after the gateway is reset from the Web or after the BURN button is clicked in the Maintenance Actions screen The number of fax calls indicated by the fields Attempted Fax Calls Counter and Successful Fax Calls Counter in the Calls Count screens may not be accurate In the screens Coders and Coder Group Settings When G 729 is used with ptimes 80 100 and 120 and G 723 is used with ptimes 120 and 150 the voice qualit
97. 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 lsProxyHotSwap 1 If the first Proxy doesn t respond to INVITE message the same INVITE message is immediately sent 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 ProxylP 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 SIP Release Notes 32 Document LTRT 65608 SIP Release Notes 1 What s New in Release 5 0 Table 1 2 Release 5 0 Modified ini File Web Parameters continues on pages 25 to 34 ini File Web Interface Parameter Name Description First Redundant Proxy IP Address Modification EnableProxySRVQuery changed to ProxyDNSQueryT ype Prox
98. 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 SIP Release Notes 68 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 Table 4 1 Release 4 8 ini File Web Browser Parameter Name continues on pages 59 to 70 ini File Web Interface Parameter Name Description AccessList_Allow_Type Action Upon Match AccessList_MatchCount A read only field that provides the number of packets accepted rejected by a Match Count specific rule PPPoE Parameters Action upon match allow or block Enables the PPPoE Point to Point Protocol over Ethernet feature EnablePPPoE 0 Disable default 1 Enable Password for PAP or Secret for CHAP authentication The valid range is a string of up to 47 characters The default value is 0 PPPoEPassword IP address to use when booting from the flash to non PPPoE Point to Point PPPoERecoverlPAddr Protocol over Ethernet environments ess The v
99. the usage of the Proxy Load Balancing mechanism Valid options include 0 Load Balancing is disabled default 1 Round Robin 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 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 RegistrarlP 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 an
100. ting transfer may hang up before the session to the transfer target is established In this scenario the transferor may wish to proceed with the transfer action Currently the gateway uses Replaces in the REFER regardless of the state of the session however this is not allowed in the Alerting state When this feature is enabled the gateway sends a CANCEL message when the transferor hangs up Thereafter the gateway waits for a 487 or 200 response to the INVITE After receiving 487 the gateway sends a REFER without Replaces However if a 200 OK response is received the gateway sends REFER with Replaces Relevant parameter EnableSemiAttendedTransfer Supports Hook Flash signaling using RFC 2833 MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports the generation of a Hook Flash signal using RFC 2833 in addition to the already supported proprietary INFO message Both options can be used simultaneously The Hook Flash signal using RFC 2833 is also indicated in the outgoing SDP Relevant parameter HookFlashOption Version 5 0 15 December 2006 a WH gA AudioCodes MediaPack Series 23 24 Supports 3 way Conference Calls MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports the creation of 3 way Conference Calls using an external media server When a specific port is involved in an active call and another call is received destined to the same port it s now possible to connect all three parti
101. tion 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 SIP Release Notes 44 Document LTRT 65608 SIP Release Notes 2 SIP Compatibility 480 481 482 483 484 485 486 487 488 Table 2 8 4xx SIP Responses continues on pages 44 to 45 4xx Response Temporarily Unavailable Call Leg Transaction Does Not Exist Loop Detected Too Many Hops Address Incomplete Ambiguous Busy Here Request Canceled Not Acceptable Supported Comments Yes Yes Yes Yes Yes Yes Yes Yes Yes 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 g
102. tor2IP gt Preference 1 20 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 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 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 5 1 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note 1 The IP ProfilelD can be used in the Tel2IP and IP2Tel routing tables Prefix and PSTNPrefix parameters Note 2 Profile Name assigned to a ProfilelD enabling User s to identify it intuitively and easily Note 3 This parameter can appear up to 9 times ID 1 to 9 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 MWlIAnalogLamp gt lt MW IDisplay gt lt FlashHookPeriod gt lt EnableEarlyMedia gt lt ProgressIndicator2 P gt
103. use Naming Authority Pointer NAPTR and or SRV DNS queries then the gateway performs resolution of that FQDN according to these mechanisms Supports NAPTR SRV resolution on multiple Contact headers in a 3xx response MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway uses DNS resolution when a 3xx response is received that includes an FQDN in the Contact header If the FQDN is resolved into more than one IP address the gateway first attempts to reach the first address If no response is received and the correct Reason for Alternative Routing is configured the gateway attempts to reach the second resolved address Version 5 0 13 December 2006 a WH wt AudioCodes MediaPack Series 16 17 18 19 Support for internal DNS SRV table MP 112 MP 114 MP 118 MP 124 FXS FXO The gateway now supports an internal DNS SRV table This table is used to resolve domain names into DNS A Records Each record in the table includes the Transport Type UDP TCP or TLS and up to three A records Each A record includes a domain name priority weight and port When an SRV resolution is required this internal DNS SRV table is searched for a match with the domain name If found the SRV record information is used if not found the query is sent to the external DNS server Relevant parameter SRVZ2IP Support for draft ietf sip gruu 09 Obtaining and Using Globally Routable User Agent UA URIs GRUU in SIP MP 112 MP 114 MP
104. w 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 Web Server itself the access level determines the extent of the access i e availability of screens and read write privileges Relevant parameter ResetWebPassword SNMP community strings can now be configured via the Embedded Web Server Relevant parameters SNMPReadOnlyCommunityString_x SNMPReadWriteCommunityString_x SNMPTrapCommunityString A new channel status screen provides the status of the R factor of RTCP XR packets per channel The Mediant 1000 chassis can now be managed via SNMP The status of the CPU and I O modules fan tray power supplies Power Entry Module PEM and other alarms are reported An HTTP download report trap is now available This trap indicates the result of a recent file download includes an HTTP error code if available Support for sysObjectID via SNMP was added Similar to MIB Object s definition Should point to a product that is defined in AC TYPES my Active analog lines performance counter is now supported in analog performance monitoring 4 5 Resolved Constraints The gateway now responds to OPTIONS messages received in mid call during a SIP dialog SIP Release Notes 58 Document LTRT 65608 SIP Release Notes 4 Previous Release 4 8 4 6 New and Modified Parameters Most new parameters des
105. y Name as Source Number Description 0 Disable default 1 Enable Keep alive with Proxy using OPTIONS 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 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 the parameter RegistrationTime Defines the time in seconds for which registration to a Proxy server is valid The value is used in the header Expires 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 is 180 seconds Determines the format of the URI in the P Assert
106. y is reduced Therefore using these ptimes isn t recommended Version 5 0 49 December 2006 Ta WH L ol AudioCodes MediaPack Series 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 The Caller ID Name column in the Caller ID table in the Embedded Web Server can t contain the inverted commas character For example entering John is not allowed In the ini file this string can be used Incorrect values are displayed on the Firewall settings Web page after the firewall rules are manipulated using SNMP The problem only occurs when mixing management interfaces i e working with the Web interface then switching to SNMP then switching back to the Web Can not access the device s Web interface using HTTPS using Microsoft Internet Explorer if DES 3DES is selected If the device is configured to use the DES cipher a logic error in Microsoft Internet Explorer would cause the HTTPS connection to fail This problem does not occur when using alternative browsers such as Firefox An ini file with parameters in table format can not be sent to a board Logo and product name revert to default with an empty ini file When Access to Restricted Domains is set to ON all attempts to enter security pages sends a Syslog message apart from CERTIFICATES page both Security certificates and SSLCertificateSR pages Parameter values set commands that
107. yIP Added NAPTR support 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 Second Redundant Proxy IP Modification EnableProxySRVQuery changed to ProxyDNSQueryType Address Added NAPTR support ProxyIP 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 Third Redundant Proxy IP Modification EnableProxySRVQuery changed to ProxyDNSQueryType Address Added NAPTR suppor

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