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Calls dropping on SIP timer expiry due to UPDATE's being ignored.
by Keith Laaks 21 Oct '09

21 Oct '09

Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name & number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? Some traces below. Any suggestions welcomed... Best Regards Keith Pretoria, South Africa. -------------------------------------------------------------------------------------------------------------------------------------------------- send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269: ------------------------------------------------------------------------ INVITE sip:27835551111@196.10.11.12 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m Max-Forwards: 67 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12> Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [calling][0] recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Content-Length: 0 ------------------------------------------------------------------------ recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [proceeding][183] 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP: v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2102 Set Codec sofia/vvrf/27835551111 G729/8000 20 ms 160 samples 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf payload to 101 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/vvrf/27835551111] 10.17.10.10 port 12862 -> 196.10.11.15 port 5230 codec: 18 ms: 20 2009-10-21 15:24:08.508561 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-10-21 15:24:08.509616 [NOTICE] sofia_glue.c:2771 Pre-Answer sofia/vvrf/27835551111! 2009-10-21 15:24:08.509616 [DEBUG] switch_channel.c:1992 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:615 Setting codec string on sofia/inetticky/27878050000(a)196.222.3.4 to G729@8000h@20i 2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:2326 sofia/inetticky/27878050000(a)196.222.3.4 receive message [PROGRESS] 2009-10-21 15:24:08.510671 [INFO] switch_ivr_originate.c:2326 Sending early media 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:0]/[G729:18:8000:20] 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2102 Set Codec sofia/inetticky/27878050000(a)196.222.3.4 G729/8000 20 ms 160 samples 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf payload to 101 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/inetticky/27878050000(a)196.222.3.4] 196.222.3.4 port 16840 -> 41.242.6.167 port 62936 codec: 18 ms: 20 2009-10-21 15:24:08.510671 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-10-21 15:24:08.511769 [NOTICE] sofia_glue.c:2771 Pre-Answer sofia/inetticky/27878050000(a)196.222.3.4! 2009-10-21 15:24:08.511769 [INFO] mod_sofia.c:1582 Ring SDP: v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.511769 [DEBUG] sofia.c:407 sofia/vvrf/27835551111 receive message [DISPLAY] 2009-10-21 15:24:08.511769 [DEBUG] switch_ivr_originate.c:2368 Originate Resulted in Success: [sofia/vvrf/27835551111] 2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182 sofia/vvrf/27835551111 receive message [AUDIO_SYNC] 2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182 sofia/inetticky/27878050000(a)196.222.3.4 receive message [AUDIO_SYNC] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:975 sofia/vvrf/27835551111 receive message [BRIDGE] 2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:982 sofia/inetticky/27878050000(a)196.222.3.4 receive message [BRIDGE] send 973 bytes to udp/[41.242.6.167]:62928 at 13:24:08.512974: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=629282009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:1026 (sofia/vvrf/27835551111) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:985 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:306 (sofia/vvrf/27835551111) Running State Change CS_EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:343 (sofia/vvrf/27835551111) State EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 INVITE Contact: <sip:0835551111@196.222.3.4:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-Actually-Support: UPDATE v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:08.512832 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [early][183] recv 928 bytes from udp/[196.10.11.12]:5060 at 13:24:14.188781: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 153288 153288 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 396 bytes to udp/[196.10.11.12]:5060 at 13:24:14.189032: ------------------------------------------------------------------------ ACK sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKZ07eFr88eH7Qg Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 ACK Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> Content-Length: 0 ------------------------------------------------------------------------ 2009-10-21 15:24:14.188910 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [ready][200] 2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:2105 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.188910 [NOTICE] sofia.c:3928 Channel [sofia/vvrf/27835551111] has been answered 2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:182 sofia/vvrf/27835551111 receive message [AUDIO_SYNC] 2009-10-21 15:24:14.204823 [DEBUG] switch_ivr_bridge.c:378 sofia/inetticky/27878050000(a)196.222.3.4 receive message [ANSWER] 2009-10-21 15:24:14.204823 [DEBUG] mod_sofia.c:543 Local SDP sofia/inetticky/27878050000(a)196.222.3.4: v=0 o=FreeSWITCH 1256114608 1256114610 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.204823 [NOTICE] switch_ivr_bridge.c:378 Channel [sofia/inetticky/27878050000(a)196.222.3.4] has been answered 2009-10-21 15:24:14.204823 [DEBUG] switch_channel.c:182 sofia/inetticky/27878050000(a)196.222.3.4 receive message [AUDIO_SYNC] send 983 bytes to udp/[41.242.6.167]:62928 at 13:24:14.205173: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=62928 From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 INVITE Contact: <sip:0835551111@196.222.3.4:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-Actually-Support: UPDATE v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:14.204823 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [completed][200] 2009-10-21 15:24:14.224959 [DEBUG] sofia.c:407 sofia/vvrf/27835551111 receive message [DISPLAY] send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.225162: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:14.224959 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [calling][0] recv 733 bytes from udp/[41.242.6.167]:62928 at 13:24:14.533488: ------------------------------------------------------------------------ ACK sip:0835551111@196.222.3.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bKd2c91e1afbf0880332fbeba5fc9c04a6;rport From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 ACK Contact: <sip:27878050000@41.242.6.167:56675> Proxy-Authorization: Digest username="27878050000", realm="196.222.3.4", qop=auth, algorithm=MD5, uri="sip:0835551111@196.222.3.4", nonce="0092f192-be45-11de-a2c1-a70e5037c20e", nc=00000001, cnonce="ab2d49987adb74997fc6b236fc16d23", response="445ae74c1f910e45d9aa1043402b4dbb" Max-Forwards: 70 User-Agent: C455 IP021910000000 Content-Length: 0 ------------------------------------------------------------------------ 2009-10-21 15:24:14.533538 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [ready][200] 2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122 sofia/vvrf/27835551111 receive message [DISPLAY] 2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122 sofia/inetticky/27878050000(a)196.222.3.4 receive message [DISPLAY] send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.727678: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:15.728255: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [Many more copies of the UPDATE's continue ... then ] 2009-10-21 15:24:46.228024 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [terminated][408] 2009-10-21 15:24:46.228024 [NOTICE] sofia.c:4039 Hangup sofia/vvrf/27835551111 [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 2009-10-21 15:24:46.228024 [DEBUG] switch_channel.c:1896 Send signal sofia/vvrf/27835551111 [KILL]

3 2

can't dial from IPv6 to IPv6
by ineya ineya 21 Oct '09

21 Oct '09

Hi, I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP port. Directory and dialplan should be default ones. My phones are able to register when using just IPv4, and I can call from one to another. When I try the same thing with IPv6, the call to other phone fails with [ USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH. Here you can find the log: http://pastebin.freeswitch.org/10757 Freeswitch is running on 10.80.62.40/2000:2::1 phones are 2000:2::1001 and 2000:2::1002 Am I doing something wrong? Maybe I forgot to configure something, but I can't see what it is. Any hints are appreciated.

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by Joey Carter 21 Oct '09

21 Oct '09

Hello How can I add background music that will play during call? Thanks

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Stop/Restart of Freeswitch Causes Crash
by Jerry Richards 21 Oct '09

21 Oct '09

Sometimes if I stop (using "..." command) and then restart freeswitch (using "./freeswitch" command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry

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Hi Guys
by Edward Q. 21 Oct '09

21 Oct '09

Hi to all of you guys (and ladies too). I would like to bother all of you and take just a little of your time. I am a total NOOB on freeswitch and would like a little help. I need to know from all of you guys which would be the hardware recommended from all of you guys to be able to deal with a DS3. The thing is i have a CANTATA switch already with the DS3 and making calls. But i will have to communicate the company international headquarters through there which in turn there are 21 offices around the world and at a certain time all of them will be passing from freeswitch server to the land lines. So could you pleae help out here. Thanks to all of you and the ones who take the time to reply. Ed PS All the calls will be comming from the web through SIP protocol into the freeswitch going out to cell phones and land lines for the executives to handle the calls. Call flow as this. A sip client on the webserver connects to the freeswitch, registers and makes the call to the support agent or the executive. Thats what we are implementing. So i guess the web server has to be another little animal to handle the traffic too. Thanks again.

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check sip client availability
by Woody Dickson 21 Oct '09

21 Oct '09

Hi, Is there any API to tell freeswitch to send a SIP OPTION message to check the availability of a SIP client? thanks, woody

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Connect PHP SOAP Web Server with SQLite database of FS
by homqua 21 Oct '09

21 Oct '09

Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your help. -- View this message in context: http://www.nabble.com/Connect-PHP-SOAP-Web-Server-with-SQLite-database-of-F… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] 2 voicemail questions
by Mark Campbell-Smith 20 Oct '09

20 Oct '09

Hi! >> 1. Can I email the voicemail message to multiple email addresses? A comma separated list does not work in the extensions.xml file (1000.xml), but it does work if I hard code the email addresses into the notify-voicemail.tpl file. Could this be added to the switch so that it can handle comma separated lists? >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition field="${vm_boxcount(${destination_number}@${domain_name})}" expression="^(1)$"> <action application="log" data="MSGBOX ${vm_boxcount(${dialed_extension}@${domain_name})}"/> <action application="set" data="api_hangup_hook=originate sofia/internal_nat/${dialed_etension}%${domain_name} default default Message 4000 4000 3"/> This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? Thanks > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? I g > From: Brian West <brian(a)freeswitch.org> > On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> > Try a comma sep. list. Not sure if it will work.

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Troubles with proxy media mode
by Kristian Kielhofner 20 Oct '09

20 Oct '09

Hello everyone, I'm trying to use proxy media across two profiles. The codec settings are identical, they both have late negotiation enabled, and they both have inbound-proxy-media set to true (I also tried setting proxy_media from the dialplan). FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but I can't figure out why it thinks it needs to transcode for this call. I've attached a level 7 debug and an ngrep siptrace showing traffic from both profiles. FreeSWITCH trunk rev. 15180 running on Debian 5.0.2. See anything interesting? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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