Hello, I'm having trouble to send/receive calls with one of my providers. 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). The sip trace follows. Anybody can help ? François send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------
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). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com 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. IPeva 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. Le 03/06/2010 à 19:36, François Legal a écrit :
Hello,
I'm having trouble to send/receive calls with one of my providers.
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).
The sip trace follows.
Anybody can help ?
François
send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419:
------------------------------------------------------------------------ INVITE sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000
recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669:
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0
------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935:
------------------------------------------------------------------------ INVITE sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp: 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453
v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20
------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380:
------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp: 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610:
------------------------------------------------------------------------ ACK sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp: 5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0
------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859:
------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0
------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043:
------------------------------------------------------------------------ ACK sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0
------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655:
------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp: 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335:
------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp: 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0
------------------------------------------------------------------------
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Get a pcap and email it to me off list... /b On Jun 3, 2010, at 12:53 PM, David Ponzone wrote:
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).
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
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. IPeva 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.
This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount@provider.com, and MyFSIpHere is the IP FS external profile is bound to. Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount@provider.com which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile. François On Thu, 3 Jun 2010 19:53:19 +0200, David Ponzone wrote: 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). David Ponzone Direction Technique email: david.ponzone@ipeva.fr [1] tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com _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. __IPEVA__ 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._ Le 03/06/2010 à 19:36, François Legal a écrit : Hello, I'm having trouble to send/receive calls with one of my providers. 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). The sip trace follows. Anybody can help ? François send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [2] 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Well looking at the invite from your provider: v=0 o=FrancoisLegal 1275590034 1275590034 IN IP4 80.239.235.114 s=SIP Call c=IN IP4 80.239.235.114 t=0 0 m=audio 11198 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11340 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 <-- THIS LINE a=rtpmap:99 <-- AND THIS LINE a=ptime:20 RFC4566 section 6: RTP profiles that specify the use of dynamic payload types MUST define the set of valid encoding names and/or a means to register encoding names if that profile is to be used with SDP. We are left guessing what 99 and 121 are which is invalid. /b On Jun 4, 2010, at 2:57 AM, François Legal wrote:
This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount@provider.com, and MyFSIpHere is the IP FS external profile is bound to.
Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount@provider.com
which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile.
François
Thanks for the help. I could fix it by setting an external profile specific outbound_codec_list not including video codecs (as I don't expect video to come from my providers) and that did the trick. François On Fri, 4 Jun 2010 09:28:35 -0500, Brian West <brian@freeswitch.org> wrote:
Well looking at the invite from your provider:
v=0 o=FrancoisLegal 1275590034 1275590034 IN IP4 80.239.235.114 s=SIP Call c=IN IP4 80.239.235.114 t=0 0 m=audio 11198 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11340 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 <-- THIS LINE a=rtpmap:99 <-- AND THIS LINE a=ptime:20
RFC4566 section 6:
RTP profiles that specify the use of dynamic payload types MUST define the set of valid encoding names and/or a means to register encoding names if that profile is to be used with SDP.
We are left guessing what 99 and 121 are which is invalid.
/b
On Jun 4, 2010, at 2:57 AM, François Legal wrote:
This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount@provider.com, and MyFSIpHere is the IP FS external profile is bound to.
Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount@provider.com
which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile.
François
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You can't fix this... the invite is coming from your provider with those in it.. and its invalid so you have no way to correct this unless you install a proxy sever and rewrite the SDP before you send it to freeswitch... your provder's VERY NICE sip proxy isn't so very NICE. /b On Jun 4, 2010, at 9:56 AM, François Legal wrote:
Thanks for the help.
I could fix it by setting an external profile specific outbound_codec_list not including video codecs (as I don't expect video to come from my providers) and that did the trick.
François
I've been seeing this more and more lately... I had to sign up for a prepaid account from a popular "provider" the other day (don't ask) and the INVITE from them advertised all kinds of crazy codecs I knew their upstreams didn't support. Stuff like G722, G726, AMR, ilbc, etc. Don't get me wrong, these are great codecs in the appropriate scenario but I *KNOW* their upstreams only support G729 and G711. Did they just not bother to change the default codec list or what? Very strange. On Fri, Jun 4, 2010 at 11:13 AM, Brian West <brian@freeswitch.org> wrote:
You can't fix this... the invite is coming from your provider with those in it.. and its invalid so you have no way to correct this unless you install a proxy sever and rewrite the SDP before you send it to freeswitch... your provder's VERY NICE sip proxy isn't so very NICE.
/b
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
participants (4)
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Brian West
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David Ponzone
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François Legal
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Kristian Kielhofner