Jeremy Salmon Posted December 28, 2011 Report Share Posted December 28, 2011 Hi All, I have a trunk to OVH SIP Provider (ovh.fr). Trunk is registered but when I place a call, I ear just the beginning of the first ring and after nothing else ... I played with trunk option, sip replacement list, ... without success snom ONE IP : 192.168.1.13 snom320 IP : 192.168.1.12 OVH sip server : 91.121.129.17 Called number : 0972101112 Here my SIP log : ===== [5] 2011/12/28 10:28:32: Last message repeated 6 times [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081: INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone> Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1 X-Serialnumber: 000413318D7F P-Key-Flags: keys="3" User-Agent: snom320/8.4.18 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 522 v=0 o=root 469490837 469490837 IN IP4 192.168.1.12 s=call c=IN IP4 192.168.1.12 t=0 0 m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [5] 2011/12/28 10:28:32: Last message repeated 2 times [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 1 INVITE Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 401 Authentication Required Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 1 INVITE User-Agent: snom-PBX/2011-4.2.0.3981 WWW-Authenticate: Digest realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",domain="sip:0972101112@societe4.topsystem.be;user=phone",algorithm=MD5 Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081: ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081: INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone> Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1 X-Serialnumber: 000413318D7F P-Key-Flags: keys="3" User-Agent: snom320/8.4.18 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="400",realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",uri="sip:0972101112@societe4.topsystem.be;user=phone",response="1f49712e3eb260cf6b2c006d031e9218",algorithm=MD5 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 522 v=0 o=root 469490837 469490837 IN IP4 192.168.1.12 s=call c=IN IP4 192.168.1.12 t=0 0 m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 INVITE Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060: INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone> Call-ID: 0f4287ed@pbx CSeq: 12053 INVITE Max-Forwards: 70 Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 323 v=0 o=- 801070725 801070725 IN IP4 192.168.1.13 s=- c=IN IP4 192.168.1.13 t=0 0 m=audio 60346 RTP/AVP 0 8 3 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 INVITE Contact: <sip:400@192.168.1.13:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 298 v=0 o=- 1825812257 1825812257 IN IP4 192.168.1.13 s=- c=IN IP4 192.168.1.13 t=0 0 m=audio 57150 RTP/AVP 0 8 3 101 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=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060: SIP/2.0 407 authentication required Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8e16178196f7d0955897860527ca4960 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736 Call-ID: 0f4287ed@pbx CSeq: 12053 INVITE Contact: <sip:0972101112@41.141.84.105:5060;user=phone> Proxy-Authenticate: Digest realm="sip.ovh.net", nonce="001a360d6b5d399f4797a37b635627ac", opaque="0010abd225acd41", stale=false, algorithm=MD5 server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060: ACK sip:0972101112@91.121.129.17;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736 Call-ID: 0f4287ed@pbx CSeq: 12053 ACK Max-Forwards: 70 Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060: INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone> Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Max-Forwards: 70 Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes Proxy-Authorization: Digest realm="sip.ovh.net",nonce="001a360d6b5d399f4797a37b635627ac",response="16890c1b4a1beaf825a0f2e3fd1234fc",username="0033184190197",uri="sip:0972101112@91.121.129.17;user=phone",opaque="0010abd225acd41",algorithm=MD5 Content-Type: application/sdp Content-Length: 323 v=0 o=- 801070725 801070725 IN IP4 192.168.1.13 s=- c=IN IP4 192.168.1.13 t=0 0 m=audio 60346 RTP/AVP 0 8 3 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081: PRACK sip:400@192.168.1.13:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 3 PRACK Max-Forwards: 70 Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1 RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 3 PRACK Contact: <sip:400@192.168.1.13:5060;transport=tcp> User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone> Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Contact: <sip:41.141.84.105:5060> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Length: 0 [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Contact: <sip:41.141.84.105:5060> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Length: 0 [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060: SIP/2.0 183 Media change Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Contact: <sip:41.141.84.105:5060> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Type: application/sdp Content-Length: 262 v=0 o=cp10 132506811328 132506811330 IN IP4 192.168.1.1 s=SIP Call c=IN IP4 192.168.1.1 t=0 0 m=audio 7072 RTP/AVP 0 8 101 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-05b06979574af555d61e176134a2f476 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1452189312 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-07824-001a1068-6632914c5 Call-ID: ad9d1f30@pbx CSeq: 18918 INVITE Contact: <sip:41.141.84.105:5060> p-asserted-identity: <sip:0972101112@91.121.129.17;user=phone> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Type: application/sdp Content-Length: 262 v=0 o=cp10 132506808498 132506808500 IN IP4 192.168.1.1 s=SIP Call c=IN IP4 192.168.1.1 t=0 0 m=audio 7070 RTP/AVP 0 8 101 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081: CANCEL sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone> Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 CANCEL Contact: <sip:400@192.168.1.13:5060;transport=tcp> User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081: SIP/2.0 487 Request Terminated Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081 From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 INVITE Contact: <sip:400@192.168.1.13:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Tx udp:91.121.129.17:5060: CANCEL sip:0972101112@91.121.129.17;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone> Call-ID: 0f4287ed@pbx CSeq: 12054 CANCEL Max-Forwards: 70 Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081: ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710 Call-ID: 3c267cc8ba57-s7m7z0rilz9z CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone> Call-ID: 0f4287ed@pbx CSeq: 12054 CANCEL server: Cirpack/v4.42j (gw_sip) Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060: SIP/2.0 487 Session canceled Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Contact: <sip:41.141.84.105:5060> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Length: 0 [5] 2011/12/28 10:28:34: SIP Tx udp:192.168.1.1:5060: ACK sip:41.141.84.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 ACK Max-Forwards: 70 Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp> Remote-Party-ID: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes Content-Length: 0 [5] 2011/12/28 10:28:34: INVITE Response 487 Session canceled: Terminate 0f4287ed@pbx [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060: ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKb094afeb726b070e8ab8f4e7222edbb0 Via: SIP/2.0/UDP 41.141.84.105:5060;branch=z9hG4bKc0e36b0d24e61351467780d2975f9369 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 ACK Contact: <sip:0033184190197@41.141.84.105> max-forwards: 68 remote-party-id: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes Content-Length: 0 [5] 2011/12/28 10:28:35: SIP Rx udp:91.121.129.17:5060: SIP/2.0 487 Session canceled Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39 Record-Route: <sip:siproxd@192.168.1.1:5060;lr> From: <sip:0033184190197@91.121.129.17>;tag=1119664557 To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090 Call-ID: 0f4287ed@pbx CSeq: 12054 INVITE Contact: <sip:41.141.84.105:5060> server: Cirpack/v4.42j (gw_sip) Allow: UPDATE, REFER, INFO Content-Length: 0 ===== Any ideas ? I thing sound disappear just after the "183 Media Change" .... Thanks in advance, Jeremy Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 28, 2011 Report Share Posted December 28, 2011 I guess the problem is that the service provider tell the PBX to send the media to "192.168.1.1". AFAIK that is a private IP address and that's where the PBX will try to send the RTP stream. Are you calling from another device in the LAN?! Does the call go through after accepting the call? Quote Link to comment Share on other sites More sharing options...
Jeremy Salmon Posted December 28, 2011 Author Report Share Posted December 28, 2011 Thanks for your response 192.168.1.1 is my gateway IP. I have 4 snom 320 in my LAN connected to my snom ONE (192.168.1.13). I can make internal call without problem. I use an another SIP Provider (ippi.fr) on this server without problem. I have just sound problem with this OVH provider. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 28, 2011 Report Share Posted December 28, 2011 Is this a smart gateway/firewall? maybe it sees the public IP address of the service provider and patches it to it's own address, so that it may relay the media. But in many cases, such "favours" cause more trouble that they help. Quote Link to comment Share on other sites More sharing options...
Jeremy Salmon Posted December 28, 2011 Author Report Share Posted December 28, 2011 It's a DLINK Dir300. But I tried with two others. Here my Trunk settings: # Trunk 3 in domain societe4.topsystem.be Name: OVH_TRUNK Type: register To: sip RegPass: ******** Direction: Disabled: Global: false Display: RegAccount: 0033184190197 RegRegistrar: sip.ovh.net RegKeep: RegUser: 0033184190197 Icid: Require: OutboundProxy: sip.ovh.net Ani: DialExtension: Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never Privacy: false Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: DialogPermission: Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 28, 2011 Report Share Posted December 28, 2011 The router might be trouble (see http://forum.snom.com/index.php?showtopic=7952). The IP address is definitevely VERY suspicious. I would have no idea who else would put this IP address there... Quote Link to comment Share on other sites More sharing options...
Jeremy Salmon Posted December 28, 2011 Author Report Share Posted December 28, 2011 For info firmware version of the DIR300 is ME_2.01. Under "Advanced Setup", "NAT" I have ALG. By default it's activated. I turned off "SIP Enabled". It seem to work fine. Thanks for your patience !!! Just a last question : why SIP Provider IPPI.fr worked and OVH no ? For info OVH have a CIRPACK and Ippi an Asterisk. Regards, JS Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 28, 2011 Report Share Posted December 28, 2011 The problem with the ALG is that the router manufacturers have to deal with so many different SIP interpretations and even buggy implementations that it becomes impossible to please everyone. SIP was not designed for ALG at all, and you need to do really difficult things to get this done properly. IMHO SIP ALG are not very useful anyway because practically all SIP providers today support a far-end NAT solution (SBC), and there is no need to fix things on the router. Sometimes it is even counter productive. Quote Link to comment Share on other sites More sharing options...
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