Re: [Freeswitch-users] Hold is broken in trunk 16055
Hold is working fine I just tested it... I would need to see the whole dialog to see what is wrong... I tested with Polycom, Snom and Aastra. Are you doing proxy media or anything like that? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96
Re: [Freeswitch-users] Hold is broken in trunk 16055
Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16
Re: [Freeswitch-users] Hold is broken in trunk 16055
Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. I will test if trunk 16055 work when I set proxy media mode to false tomorrow. 2009/12/29 Brian West br...@freeswitch.org Hold is working fine I just tested it... I would need to see the whole dialog to see what is wrong... I tested with Polycom, Snom and Aastra. Are you doing proxy media or anything like that? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer,
Re: [Freeswitch-users] Hold is broken in trunk 16055
The phone I'm using is x-lite and wxCommunicator, both are sip phone software. I have not used pastebin, Is it a bug trace tool like bugzilla? Can you tell me how to register a pastbin account? 2009/12/29 Brian West br...@freeswitch.org Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120
Re: [Freeswitch-users] Hold is broken in trunk 16055
the 200ok is not from FS.. its from the end point... so its not us thats not putting the SDP into the 200ok but the device you're talking to because in proxy media they are passed as is. /b On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. I will test if trunk 16055 work when I set proxy media mode to false tomorrow. ___ 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
Re: [Freeswitch-users] Hold is broken in trunk 16055
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
else if (tech_pvt sofia_test_flag(tech_pvt, TFLAG_SDP)
!r_sdp) {
nua_respond(tech_pvt-nh, SIP_200_OK, TAG_END());
sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
goto done;
}
The cause is r_sdp is null, but I don't known why tl_gets don't return
remote sdp tag, it's quite strange.
2009/12/29 Brian West br...@freeswitch.org
the 200ok is not from FS.. its from the end point... so its not us thats
not putting the SDP into the 200ok but the device you're talking to because
in proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy media mode. the
sip agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to false
tomorrow.
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Re: [Freeswitch-users] Hold is broken in trunk 16055
Btw, in the same scenario, FS 1.0.4 works fine.
2009/12/29 Lei Tang lei.tl...@gmail.com
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
else if (tech_pvt sofia_test_flag(tech_pvt, TFLAG_SDP)
!r_sdp) {
nua_respond(tech_pvt-nh, SIP_200_OK, TAG_END());
sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
goto done;
}
The cause is r_sdp is null, but I don't known why tl_gets don't return
remote sdp tag, it's quite strange.
2009/12/29 Brian West br...@freeswitch.org
the 200ok is not from FS.. its from the end point... so its not us thats
not putting the SDP into the 200ok but the device you're talking to because
in proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy media mode. the
sip agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to false
tomorrow.
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Re: [Freeswitch-users] Hold is broken in trunk 16055
There is no need for you to show us traces. The fact that you are using proxy media is enough to know that the issue is with your device. If you look at the full sip trace you will see the same. Mike On Dec 29, 2009, at 10:10 AM, Lei Tang lei.tl...@gmail.com wrote: The phone I'm using is x-lite and wxCommunicator, both are sip phone software. I have not used pastebin, Is it a bug trace tool like bugzilla? Can you tell me how to register a pastbin account? 2009/12/29 Brian West br...@freeswitch.org Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO,
Re: [Freeswitch-users] Hold is broken in trunk 16055
Its null because the device on the other side didn't send one. We pass it as
is... fix the broken device or don't use proxy media.
/b
On Dec 29, 2009, at 9:37 AM, Lei Tang wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
else if (tech_pvt sofia_test_flag(tech_pvt, TFLAG_SDP)
!r_sdp) {
nua_respond(tech_pvt-nh, SIP_200_OK, TAG_END());
sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
goto done;
}
The cause is r_sdp is null, but I don't known why tl_gets don't return remote
sdp tag, it's quite strange.
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Re: [Freeswitch-users] Hold is broken in trunk 16055
This means there was no sdp sent. Did you confirm this with siptrace?
On Dec 29, 2009, at 10:37 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong,
following code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
else if (tech_pvt sofia_test_flag(tech_pvt,
TFLAG_SDP) !r_sdp) {
nua_respond(tech_pvt-nh, SIP_200_OK, TAG_END());
sofia_set_flag_locked(tech_pvt,
TFLAG_NOSDP_REINVITE);
goto done;
}
The cause is r_sdp is null, but I don't known why tl_gets don't
return remote sdp tag, it's quite strange.
2009/12/29 Brian West br...@freeswitch.org
the 200ok is not from FS.. its from the end point... so its not us
thats not putting the SDP into the 200ok but the device you're
talking to because in proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy media mode.
the sip agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to
false tomorrow.
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Re: [Freeswitch-users] Hold is broken in trunk 16055
We now disable sofia SOA mode during proxy calls.
This means that sofia will not try to get involved in the media negotiation
at all which is the optimal behavior.
Previous versions would butt in and try to fix the error but now it just
stays out of the way.
You can see in your trace that the device sends a packet with no SDP
therefore so does sofia.
You can either turn off proxy-media or post a bounty for me to go hack a
workaround into the patch I spent many hours on getting things to work
right. Whatever you experienced with 1.0.4 was a happy coincidence where
sofia was fixing a bug in your phone for you.
On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris m...@jerris.com wrote:
This means there was no sdp sent. Did you confirm this with siptrace?
On Dec 29, 2009, at 10:37 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
else if (tech_pvt sofia_test_flag(tech_pvt, TFLAG_SDP)
!r_sdp) {
nua_respond(tech_pvt-nh, SIP_200_OK, TAG_END());
sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
goto done;
}
The cause is r_sdp is null, but I don't known why tl_gets don't return
remote sdp tag, it's quite strange.
2009/12/29 Brian West br...@freeswitch.orgbr...@freeswitch.org
the 200ok is not from FS.. its from the end point... so its not us thats
not putting the SDP into the 200ok but the device you're talking to because
in proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy media mode. the
sip agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to false
tomorrow.
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[Freeswitch-users] Hold is broken in trunk 16055
Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK
