<p>This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach <a>SomeAccount@provider.com, and MyFSIpHere is the IP FS external profile is bound to.</a></p>
<p>Maybe this lack some information : the "dialed number" is originate sofia/external/<a>SomeAccount@provider.com</a></p>
<p>which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile.</p>
<p> </p>
<p>François</p>
<p>On Thu, 3 Jun 2010 19:53:19 +0200, David Ponzone <david.ponzone@gmail.com> wrote:</p>
<blockquote type="cite" style="padding-left:5px; border-left:#1010ff 2px solid; margin-left:5px; width:100%"><!-- html ignored --><!-- head ignored --><!-- meta ignored -->I would really recommend you resend the traces with the full headers (change the phone numbers and IPs if you need to, but replace them by variables so that we know when the same IP or number is present in various fields/headers).
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<div><span style="font-family: 'Helvetica Neue';"><span style="color: #1c00ff;">David Ponzone </span><span style="color: #000000; font-size: small;"><span class="Apple-style-span" style="font-size: 12px;">Direction Technique</span></span></span></div>
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<div style="text-align: justify; font: normal normal normal 10px; color: #c0c0c0; margin: 0px;"><em>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </em><strong><em>IPeva</em></strong><em> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</em></div>
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<div>Le 03/06/2010 à 19:36, François Legal a écrit :</div>
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<blockquote type="cite" style="padding-left:5px; border-left:#1010ff 2px solid; margin-left:5px; width:100%">
<p>Hello,</p>
<div> <br class="webkit-block-placeholder" /></div>
<p>I'm having trouble to send/receive calls with one of my providers.</p>
<p>Freeswitch says the reply sent by provider has the session information incorrect, then the provider replies the calle is busy (which of course is not).</p>
<div> <br class="webkit-block-placeholder" /></div>
<p>The sip trace follows.</p>
<div> <br class="webkit-block-placeholder" /></div>
<p>Anybody can help ?</p>
<div> <br class="webkit-block-placeholder" /></div>
<p>François</p>
<div> <br class="webkit-block-placeholder" /></div>
<p>send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419:<br /> ------------------------------------------------------------------------<br /> INVITE <a>sip:SomeAccount@provider.com</a> SIP/2.0<br /> Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK<br /> Max-Forwards: 70<br /> From: "" ;tag=2SU9FHayaSXcr<br /> To: <br /> Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e<br /> CSeq: 131671088 INVITE<br /> Contact: <br /> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-<br /> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br /> Supported: timer, precondition, path, replaces<br /> Allow-Events: talk, hold, refer<br /> Content-Type: application/sdp<br /> Content-Disposition: session<br /> Content-Length: 642<br /> X-FS-Support: update_display<br /> Remote-Party-ID: ;party=calling;screen=yes;privacy=off<br /> <br /> v=0<br /> o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere<br /> s=FreeSWITCH<br /> c=IN IP4 MyFSIpHere<br /> t=0 0<br /> m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13<br /> a=rtpmap:124 G726-16/8000<br /> a=rtpmap:123 G726-24/8000<br /> a=rtpmap:122 G726-32/8000<br /> a=rtpmap:121 G726-40/8000<br /> a=rtpmap:3 GSM/8000<br /> a=rtpmap:99 SPEEX/8000<br /> a=rtpmap:9 G722/8000<br /> a=rtpmap:8 PCMA/8000<br /> a=rtpmap:0 PCMU/8000<br /> a=rtpmap:101 telephone-event/8000<br /> a=fmtp:101 0-16<br /> a=rtpmap:13 CN/8000<br /> a=ptime:20<br /> m=video 19186 RTP/AVP 31 34 115 121 99<br /> a=rtpmap:31 H261/90000<br /> a=rtpmap:34 H263/90000<br /> a=rtpmap:115 H263-1998/90000<br /> a=rtpmap:121 H263-2000/90000<br /> a=rtpmap:99 H264/90000<br /><br />recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669:<br /> ------------------------------------------------------------------------<br /> SIP/2.0 100 Trying<br /> Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK<br /> From: "" ;tag=2SU9FHayaSXcr<br /> To: <br /> Contact: <a>sip:MyProviderIp:5060</a><br /> Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e<br /> CSeq: 131671088 INVITE<br /> Server: (Very nice Sip Registrar/Proxy Server)<br /> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935:<br /> ------------------------------------------------------------------------<br /> INVITE <a>sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com</a> SIP/2.0<br /> Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48<br /> From: "0000000000" ;tag=4e0113ac4c050c86c51a2<br /> To: <br /> Contact: <a>sip:0000000000@MyProviderIp:5060</a><br /> Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp<br /> CSeq: 1 INVITE<br /> User-Agent: (Very nice Sip Registrar/Proxy Server)<br /> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE<br /> Content-Type: application/sdp<br /> Content-Length: 453<br /> <br /> v=0<br /> o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106<br /> s=SIP Call<br /> c=IN IP4 80.239.235.106<br /> t=0 0<br /> m=audio 11180 RTP/AVP 2 3 9 8 0 101<br /> a=rtpmap:2 G726-32/8000<br /> a=rtpmap:3 GSM/8000<br /> a=rtpmap:9 G722/8000<br /> a=rtpmap:8 PCMA/8000<br /> a=rtpmap:0 PCMU/8000<br /> a=rtpmap:101 telephone-event/8000<br /> m=video 11598 RTP/AVP 31 34 115 121 99<br /> a=rtpmap:31 H261/90000<br /> a=rtpmap:34 H263/90000<br /> a=rtpmap:115 H263-1998/90000<br /> a=rtpmap:121 <br /> a=rtpmap:99 <br /> a=ptime:20<br /> ------------------------------------------------------------------------<br />send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380:<br /> ------------------------------------------------------------------------<br /> SIP/2.0 400 Bad Session Description<br /> Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48<br /> From: "0000000000" ;tag=4e0113ac4c050c86c51a2<br /> To: ;tag=32m2HcU171KZK<br /> Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp<br /> CSeq: 1 INVITE<br /> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-<br /> Accept: application/sdp<br /> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br /> Supported: timer, precondition, path, replaces<br /> Allow-Events: talk, hold, refer<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610:<br /> ------------------------------------------------------------------------<br /> ACK <a>sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com</a> SIP/2.0<br /> Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0<br /> From: "0000000000" ;tag=4e0113ac4c050c86c51a2<br /> To: ;tag=32m2HcU171KZK<br /> Contact: <a>sip:0000000000@MyProviderIp:5060</a><br /> Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp<br /> CSeq: 1 ACK<br /> Server: (Very nice Sip Registrar/Proxy Server)<br /> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859:<br /> ------------------------------------------------------------------------<br /> SIP/2.0 486 Busy here<br /> Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK<br />freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr<br /> To: <br /> Contact: <a>sip:MyProviderIp:5060</a><br /> Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e<br /> CSeq: 131671088 INVITE<br /> Server: (Very nice Sip Registrar/Proxy Server)<br /> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043:<br /> ------------------------------------------------------------------------<br /> ACK <a>sip:SomeAccount@provider.com</a> SIP/2.0<br /> Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK<br /> Max-Forwards: 70<br /> From: "" ;tag=2SU9FHayaSXcr<br /> To: <br /> Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e<br /> CSeq: 131671088 ACK<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655:<br /> ------------------------------------------------------------------------<br /> SIP/2.0 400 Bad Session Description<br /> Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48<br /> From: "0000000000" ;tag=4e0113ac4c050c86c51a2<br /> To: ;tag=32m2HcU171KZK<br /> Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp<br /> CSeq: 1 INVITE<br /> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-<br /> Accept: application/sdp<br /> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br /> Supported: timer, precondition, path, replaces<br /> Allow-Events: talk, hold, refer<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br />recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335:<br /> ------------------------------------------------------------------------<br /> ACK <a>sip:77.72.169.134:5060</a> SIP/2.0<br /> Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48<br /> From: "0000000000" ;tag=4e0113ac4c050c86c51a2<br /> To: ;tag=32m2HcU171KZK<br /> Contact: <a>sip:77.72.169.134:5060</a><br /> Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp<br /> CSeq: 1 ACK<br /> Server: (Very nice Sip Registrar/Proxy Server)<br /> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE<br /> Content-Length: 0<br /> <br /> ------------------------------------------------------------------------<br /><br /><br /><br /><br /><br /><br /><br /></p>
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