The problem has been fixed. We are able to hear audio in our calls after adding these lines in the AS5300 config:
sip-ua g729-annexb override There's an issue regarding codec matching in IOS versions 12.3(18) or higher: https://supportforums.cisco.com/docs/DOC-3186 On Tue, Jan 10, 2012 at 10:30 AM, Roi Stork <[email protected]> wrote: > > Here's the cisco AS5300 settings from our provider > > codec preference 1 g729r8 > codec preference 2 g729br8 > codec preference 3 g723r53 > codec preference 4 g723r63 > codec preference 5 g723ar53 > codec preference 6 g723ar63 > > On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork <[email protected]> wrote: >> >> Hi Alex, here's the config and the sip debug output. >> >> Guide: >> xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add >> yyy.yy.yy.yy - our asterisk 1.6.2.14 server >> >> sip config: >> >> type=peer >> disallow=all >> allow=g729 >> host=xxx.xxx.xxx.xxx >> fromdomain=xxx.xxx.xxx.xxx >> dtmfmode=rfc2833 >> nat=no >> canreinvite=yes >> context=from-trunk-sip-iaccess >> >> sip debug: >> v=0 >> o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy >> s=Asterisk PBX 1.6.2.14 >> c=IN IP4 yyy.yy.yy.yy >> t=0 0 >> m=audio 13702 RTP/AVP 0 8 3 18 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> --- >> >> <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport >> From: "6598715968" <sip:[email protected]>;tag=as6e218907 >> To: <sip:[email protected]> >> Date: Fri, 06 Jan 2012 04:51:39 GMT >> Call-ID: [email protected] >> Server: Cisco-SIPGateway/IOS-12.x >> CSeq: 102 INVITE >> Allow-Events: telephone-event >> Content-Length: 0 >> >> >> <-------------> >> --- (10 headers 0 lines) --- >> Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060: >> OPTIONS sip:zzz.zz.zz.zz SIP/2.0 >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport >> Max-Forwards: 70 >> From: "Unknown" <sip:[email protected]>;tag=as5c8e3f97 >> To: <sip:zzz.zz.zz.zz> >> Contact: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 1.6.2.14 >> Date: Fri, 06 Jan 2012 06:23:00 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:69.90.209.57:5060 ---> >> >> <-------------> >> Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060: >> OPTIONS sip:zzz.zz.zz.zz SIP/2.0 >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport >> Max-Forwards: 70 >> From: "Unknown" <sip:[email protected]>;tag=as5c8e3f97 >> To: <sip:zzz.zz.zz.zz> >> Contact: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 1.6.2.14 >> Date: Fri, 06 Jan 2012 06:23:00 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Really destroying SIP dialog '[email protected]' >> Method: OPTIONS >> >> <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> >> SIP/2.0 183 Session Progress >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport >> From: "6598715968" <sip:[email protected]>;tag=as6e218907 >> To: <sip:[email protected]>;tag=B6534850-EC6 >> Date: Fri, 06 Jan 2012 04:51:39 GMT >> Call-ID: [email protected] >> Server: Cisco-SIPGateway/IOS-12.x >> CSeq: 102 INVITE >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, >> NOTIFY, INFO, REGISTER >> Allow-Events: telephone-event >> Remote-Party-ID: "6598715968" >> >> <sip:1234#[email protected]>;party=called;screen=no;privacy=off >> Contact: <sip:[email protected]:5060> >> Content-Type: application/sdp >> Content-Disposition: session;handling=required >> Content-Length: 223 >> >> v=0 >> o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx >> s=SIP Call >> c=IN IP4 xxx.xxx.xxx.xxx >> t=0 0 >> m=audio 18132 RTP/AVP 18 >> c=IN IP4 xxx.xxx.xxx.xxx >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=ptime:20 >> >> <-------------> >> --- (15 headers 10 lines) --- >> Found RTP audio format 18 >> Found audio description format G729 for ID 18 >> Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 >> (g729)/video=0x0 >> >> (nothing)/text=0x0 (nothing), combined - 0x100 (g729) >> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 >> (nothing), combined - 0x0 >> >> (nothing) >> Peer audio RTP is at port xxx.xxx.xxx.xxx:18132 >> >> <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport >> From: "6598715968" <sip:[email protected]>;tag=as6e218907 >> To: <sip:[email protected]>;tag=B6534850-EC6 >> Date: Fri, 06 Jan 2012 04:51:39 GMT >> Call-ID: [email protected] >> Server: Cisco-SIPGateway/IOS-12.x >> CSeq: 102 INVITE >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, >> NOTIFY, INFO, REGISTER >> Allow-Events: telephone-event >> Contact: <sip:[email protected]:5060> >> Supported: replaces >> Content-Type: application/sdp >> Content-Disposition: session;handling=required >> Content-Length: 223 >> >> v=0 >> o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx >> s=SIP Call >> c=IN IP4 xxx.xxx.xxx.xxx >> t=0 0 >> m=audio 18132 RTP/AVP 18 >> c=IN IP4 xxx.xxx.xxx.xxx >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=ptime:20 >> >> <-------------> >> --- (15 headers 10 lines) --- >> list_route: hop: <sip:[email protected]:5060> >> set_destination: Parsing <sip:[email protected]:5060> for >> address/port to send to >> set_destination: set destination to xxx.xxx.xxx.xxx, port 5060 >> Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: >> ACK sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK17854b94;rport >> Max-Forwards: 70 >> From: "6598715968" <sip:[email protected]>;tag=as6e218907 >> To: <sip:[email protected]>;tag=B6534850-EC6 >> Contact: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 ACK >> User-Agent: Asterisk PBX 1.6.2.14 >> Content-Length: 0 >> >> >> --- >> > Channel SIP/xxx.xxx.xxx.xxx-00003693 was answered. >> -- Executing [6591394459@a2billing-callback:1] >> DeadAGI("SIP/xxx.xxx.xxx.xxx-00003693", >> >> "a2billing.php,1,callback") in new stack >> -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php >> -- AGI Script Executing Application: (DIAL) Options: >> >> (SIP/xxx.xxx.xxx.xxx/34546591394459,60,HRrL(370239000:61000:30000)) >> -- Limit Data for this call: >> > timelimit = 370239000 >> > play_warning = 61000 >> > play_to_caller = yes >> > play_to_callee = no >> > warning_freq = 30000 >> > start_sound = >> > warning_sound = timeleft >> > end_sound = >> == Using SIP RTP TOS bits 184 >> == Using SIP RTP CoS mark 5 >> Audio is at yyy.yy.yy.yy port 14212 >> Adding codec 0x100 (g729) to SDP >> Adding codec 0x4 (ulaw) to SDP >> Adding codec 0x8 (alaw) to SDP >> Adding codec 0x2 (gsm) to SDP >> Adding non-codec 0x1 (telephone-event) to SDP >> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: >> INVITE sip:[email protected] SIP/2.0 >> Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK4ea95f20;rport >> Max-Forwards: 70 >> From: "6598715968" <sip:[email protected]>;tag=as492477b7 >> To: <sip:[email protected]> >> Contact: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX 1.6.2.14 >> Date: Fri, 06 Jan 2012 06:23:10 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> Supported: replaces, timer >> Content-Type: application/sdp >> Content-Length: 331 >> >> v=0 >> o=root 1686167830 1686167830 IN IP4 yyy.yy.yy.yy >> s=Asterisk PBX 1.6.2.14 >> c=IN IP4 yyy.yy.yy.yy >> t=0 0 >> m=audio 14212 RTP/AVP 18 0 8 3 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> >> >> To: <sip:[email protected]>;tag=B6534850-EC6 >> Date: Fri, 06 Jan 2012 04:51:39 GMT >> Call-ID: [email protected] >> Server: Cisco-SIPGateway/IOS-12.x >> CSeq: 102 INVITE >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, >> NOTIFY, INFO, REGISTER >> Allow-Events: telephone-event >> Remote-Party-ID: "6598715968" >> >> <sip:1234#[email protected]>;party=called;screen=no;privacy=off >> Contact: <sip:[email protected]:5060> >> Content-Type: application/sdp >> Content-Disposition: session;handling=required >> Content-Length: 223 >> >> v=0 >> o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx >> s=SIP Call >> c=IN IP4 xxx.xxx.xxx.xxx >> t=0 0 >> m=audio 18132 RTP/AVP 18 >> c=IN IP4 xxx.xxx.xxx.xxx >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=ptime:20 >> >> >> >> On Mon, Jan 9, 2012 at 4:33 PM, Alex Balashov <[email protected]> >> wrote: >>> >>> You are hereby encouraged to post your AS5300 IOS config, sip.conf peer >>> declaration, and packet capture. Those three things would aid greatly in >>> diagnosis, especially the capture. >>> >>> -- >>> This message was painstakingly thumbed out on my mobile, so apologies for >>> brevity, errors, and general sloppiness. >>> >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 260 Peachtree Street NW >>> Suite 2200 >>> Atlanta, GA 30303 >>> Tel: +1-678-954-0670 >>> Fax: +1-404-961-1892 >>> Web: http://www.evaristesys.com/ >>> >>> On Jan 9, 2012, at 3:20 AM, Roi Stork <[email protected]> wrote: >>> >>> > Hi, >>> > >>> > We have a problem connecting to a Cisco AS5300 trunk. >>> > >>> > We set the sip peer to allow only g729. The call attempt is able to >>> > connect, but when answered, no audio is heard or transmitted. >>> > >>> > Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium. >>> > >>> > We do not have this problem on our other providers using asterisk and >>> > other non-cisco systems. >>> > Anyone else having this same problem? >>> > -- >>> > _____________________________________________________________________ >>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> > New to Asterisk? Join us for a live introductory webinar every Thurs: >>> > http://www.asterisk.org/hello >>> > >>> > asterisk-users mailing list >>> > To UNSUBSCRIBE or update options visit: >>> > http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
