ser*CLI> <--- SIP read from UDP:ip1:5060 ---> INVITE sip:test@ip1:5065 SIP/2.0 Record-Route: Via: SIP/2.0/UDP ip1;branch=z9hG4bK9a1.e5326d92.0 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bKce7cdbc0324b5c39c1c5241653fcf4a1.0 Session-Expires: 10800 From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 2 INVITE Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27558-99a6365-491ad47c.ju+2kRTPk--AGzBDvCuSwg__ Expires: 180 Max-Forwards: 68 Supported: 100rel,replaces User-Agent: eConf4.2.71 Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain Contact: Content-Type: application/sdp Content-Length: 614 Record-Route: P-hint: outbound v=0 o=anonymous 1261127079 1261127079 IN IP4 ip2 s=- i=eConf4.2.71 c=IN IP4 ip2 b=AS:384 t=0 0 m=audio 58756 RTP/AVP 8 0 18 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no m=video 58758 RTP/AVP 97 34 31 b=TIAS:384000 b=AS:384 a=rtpmap:97 H263-1998/90000 a=maxprate:40.0 a=fmtp:97 CIF=1;QCIF=1;I=1;J=1;T=1;N=4;K=1 a=sendrecv a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1;QCIF=1 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 m=text 58760 RTP/AVP 99 98 a=rtpmap:99 RED/1000 a=rtpmap:98 T140/1000 a=fmtp:99 98/98/98 a=sendrecv <-------------> --- (20 headers 29 lines) --- Sending to ip1 : 5060 (no NAT) Using INVITE request as basis request - 17f4588-0-13c4-40030-27558-52e363d3-27558 Found peer 'peter' for 'peter' from ip1:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found RTP video format 97 Found RTP video format 34 Found RTP video format 31 Found video description format H263-1998 for ID 97 Found video description format H263 for ID 34 Found video description format H261 for ID 31 Found RTP text format 99 Found RTP text format 98 RED submimetype has payload type: 99 Capabilities: us - 0xc08000c (ulaw|alaw|h263|red|t140), peer - audio=0x10c (ulaw|alaw|g729)/video=0x1c0000 (h261|h263|h263p)/text=0xc000000 (red|t140), combined - 0xc08000c (ulaw|alaw|h263|red|t140) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port ip2:58756 Peer video RTP is at port ip2:58758 Peer T.140 RTP is at port ip2:58758 Looking for test in default (domain ip1) list_route: hop: list_route: hop: <--- Transmitting (no NAT) to ip1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP ip1;branch=z9hG4bK9a1.e5326d92.0;received=ip1 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bKce7cdbc0324b5c39c1c5241653fcf4a1.0 Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27558-99a6365-491ad47c.ju+2kRTPk--AGzBDvCuSwg__ Record-Route: Record-Route: From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r232580M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at ip1 port 13312 Video is at ip1 port 10396 We think we can do text And we have a text rtp object Lets set up the text sdp Text is at ip1 port 19238 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding video codec 0x80000 (h263) to SDP Adding text codec 0x4000000 (red) to SDP Adding text codec 0x8000000 (t140) to SDP <--- Reliably Transmitting (no NAT) to ip1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip1;branch=z9hG4bK9a1.e5326d92.0;received=ip1 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bKce7cdbc0324b5c39c1c5241653fcf4a1.0 Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27558-99a6365-491ad47c.ju+2kRTPk--AGzBDvCuSwg__ Record-Route: Record-Route: From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: ;tag=as3c910114 Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r232580M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 421 v=0 o=root 100388057 100388057 IN IP4 ip1 s=Asterisk PBX SVN-trunk-r232580M c=IN IP4 ip1 b=CT:384 t=0 0 m=audio 13312 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 10396 RTP/AVP 34 a=rtpmap:34 H263/90000 a=sendrecv m=text 19238 RTP/AVP 99 98 a=rtpmap:99 RED/1000 a=fmtp:99 98/98/98 a=rtpmap:98 T140/1000 a=sendrecv <------------> gsm seek returned: 0We have T.140 We dont have FD <--- SIP read from UDP:ip1:5060 ---> ACK sip:test@ip1:5065 SIP/2.0 Via: SIP/2.0/UDP ip1;branch=z9hG4bK9a1.e5326d92.2 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bK68aeb8dfd1f938ba3d1eb7e387b334c2.0 From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: ;tag=as3c910114 Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 2 ACK Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27559-99a66f3-38503df5.AsjUMJ5Ollhnz8Gx+phYCw__ Max-Forwards: 68 User-Agent: eConf4.2.71 Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain Contact: Proxy-Authorization: Digest username="peter",realm="domain.com",nonce="4b2b45c00000001142e48f38f3fc94d23714e09646e0b6b6",uri="sip:test@domain.com",response="3eeb0e044fd356c9c9b9896e9b7163b1",algorithm=MD5 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to ip1, port 5060 Reliably Transmitting (no NAT) to ip1:5060: INFO sip:sip%3apeter%40127.0.0.1@ip2 SIP/2.0 Via: SIP/2.0/UDP ip1:5065;branch=z9hG4bK113f1424 Route: , Max-Forwards: 70 From: ;tag=as3c910114 To: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 Contact: Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 102 INFO User-Agent: Asterisk PBX SVN-trunk-r232580M Content-Type: application/media_control+xml Content-Length: 205 --- ser*CLI> <--- SIP read from UDP:ip1:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP ip1:5065;rport=5065;received=ip1;branch=z9hG4bK113f1424 Max-Forwards: 67 From: ;tag=as3c910114 To: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 Contact: Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 102 INFO User-Agent: Asterisk PBX SVN-trunk-r232580M Content-Type: application/media_control+xml Content-Length: 205 <-------------> --- (11 headers 9 lines) --- SIP Response message for INCOMING dialog INFO arrived ser*CLI> <--- SIP read from UDP:ip1:5060 ---> BYE sip:test@ip1:5065 SIP/2.0 Via: SIP/2.0/UDP ip1;branch=z9hG4bKaa1.03e611f7.0 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bKb9023c468edea6e53d1eb7e387b334c2.0 From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: ;tag=as3c910114 Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 3 BYE Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27577-99adb43-1f07b56b.Qz3xBOosVGRnfLXEHaNfow__ Max-Forwards: 68 Supported: 100rel,replaces User-Agent: eConf4.2.71 Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain Proxy-Authorization: Digest username="peter",realm="domain.com",nonce="4b2b45c00000001142e48f38f3fc94d23714e09646e0b6b6",uri="sip:test@ip1:5065",response="df8c71cc9db59c971a2c20160a39bb31",algorithm=MD5 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to ip1 : 5060 (no NAT) ser*CLI> <--- Transmitting (no NAT) to ip1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip1;branch=z9hG4bKaa1.03e611f7.0;received=ip1 Via: SIP/2.0/UDP ip2:5060;rport=5060;received=ip2;branch=z9hG4bKb9023c468edea6e53d1eb7e387b334c2.0 Via: SIP/2.0/UDP ip2:5060;branch=z9hG4bK-27577-99adb43-1f07b56b.Qz3xBOosVGRnfLXEHaNfow__ From: "peter";tag=17ed2d0-0-13c4-40030-27558-29ca5b29-27558 To: ;tag=as3c910114 Call-ID: 17f4588-0-13c4-40030-27558-52e363d3-27558 CSeq: 3 BYE Server: Asterisk PBX SVN-trunk-r232580M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Dec 18 10:05:05] WARNING[7635]: format_pcm.c:124 pcm_seek: negative offset -2000, resetting to 0 Really destroying SIP dialog '17f4588-0-13c4-40030-27558-52e363d3-27558' Method: BYE ser*CLI> root@ser:~#