<--- SIP read from UDP://10.0.11.1:1221 ---> INVITE sip:1593@192.168.51.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKe10e5decd Max-Forwards: 70 Content-Length: 565 To: citywok SVR1 From: citywok SVR1 ;tag=5230473a2e5a566 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797645 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: citywok SVR1 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1564010590 IN IP4 192.168.16.221 s=SIP Call c=IN IP4 192.168.16.221 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> csgtacsip1*CLI> --- (15 headers 23 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.16.221 : 5060 (no NAT) Using INVITE request as basis request - 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 No user '1593' in SIP users list Found peer '1593' for '1593' from 10.0.11.1:1221 <--- Reliably Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKe10e5decd;received=10.0.11.1 From: citywok SVR1 ;tag=5230473a2e5a566 To: citywok SVR1 ;tag=as76b34f66 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797645 INVITE User-Agent: Asterisk PBX 1.6.0.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="471d56f1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '33e7c9839c51a59b871b2157e53c4c69@192.168.16.221' in 7104 ms (Method: INVITE) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> ACK sip:1593@192.168.51.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKe10e5decd Max-Forwards: 70 Content-Length: 0 To: citywok SVR1 ;tag=as76b34f66 From: citywok SVR1 ;tag=5230473a2e5a566 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797645 ACK User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> INVITE sip:1593@192.168.51.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKa9a8472a2 Max-Forwards: 70 Content-Length: 565 To: citywok SVR1 From: citywok SVR1 ;tag=5230473a2e5a566 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797646 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: citywok SVR1 Content-Type: application/sdp Supported: replaces Authorization:Digest response="6f6eabe0abe2b90e2ec3aae089d35d80",username="1593",realm="asterisk",nonce="471d56f1",algorithm=MD5,uri="sip:1593@192.168.51.15:5060" User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1564010590 IN IP4 192.168.16.221 s=SIP Call c=IN IP4 192.168.16.221 t=0 0 m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> --- (16 headers 23 lines) --- Sending to 10.0.11.1 : 1221 (NAT) Using INVITE request as basis request - 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 No user '1593' in SIP users list csgtacsip1*CLI> Found peer '1593' for '1593' from 10.0.11.1:1221 csgtacsip1*CLI> Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.16.221:3000 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found unknown media description format BV16 for ID 96 Found unknown media description format BV32 for ID 102 Found audio description format L16 for ID 107 Found audio description format PCMU for ID 104 Found audio description format PCMA for ID 105 Found audio description format L16 for ID 106 Found unknown media description format G726-16 for ID 97 Found unknown media description format G726-24 for ID 98 Found audio description format G726-32 for ID 2 Found unknown media description format G726-40 for ID 99 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x94c (ulaw|alaw|g726|slin|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.16.221:3000 Looking for 1593 in from-staff (domain 192.168.51.15) list_route: hop: <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKa9a8472a2;received=10.0.11.1 From: citywok SVR1 ;tag=5230473a2e5a566 To: citywok SVR1 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797646 INVITE User-Agent: Asterisk PBX 1.6.0.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> csgtacsip1*CLI> -- Executing [1593@from-staff:1] Dial("SIP/1593-08f5a370", "SIP/OCS_TRUNK/+2593") in new stack == Using SIP RTP CoS mark 5 csgtacsip1*CLI> Audio is at 192.168.51.15 port 15714 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP csgtacsip1*CLI> Reliably Transmitting (no NAT) to 192.168.51.16:5060: INVITE sip:+2593@192.168.51.16 SIP/2.0 Via: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport Max-Forwards: 70 From: "citywok" ;tag=as23a917d8 To: Contact: Call-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.13 Date: Fri, 09 Oct 2009 17:54:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1798051224 1798051224 IN IP4 192.168.51.15 s=Asterisk PBX 1.6.0.13 c=IN IP4 192.168.51.15 t=0 0 m=audio 15714 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- csgtacsip1*CLI> -- Called OCS_TRUNK/+2593 csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 100 Trying FROM: "citywok";tag=as23a917d8 TO: CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 <-------------> csgtacsip1*CLI> --- (7 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 183 Session Progress FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 Ringing FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> -- SIP/OCS_TRUNK-f7628128 is ringing csgtacsip1*CLI> <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKa9a8472a2;received=10.0.11.1 From: citywok SVR1 ;tag=5230473a2e5a566 To: citywok SVR1 ;tag=as268797a3 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797646 INVITE User-Agent: Asterisk PBX 1.6.0.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 Ringing FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> -- SIP/OCS_TRUNK-f7628128 is ringing csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 Ringing FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> -- SIP/OCS_TRUNK-f7628128 is ringing csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 Ringing FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> -- SIP/OCS_TRUNK-f7628128 is ringing csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 200 OK FROM: "citywok";tag=as23a917d8 TO: ;epid=FCE3547589;tag=c296554458 CSEQ: 102 INVITE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK0eeeef1a;rport CONTACT: CONTENT-LENGTH: 256 SUPPORTED: 100rel CONTENT-TYPE: application/sdp; charset=utf-8 ALLOW: UPDATE SERVER: RTCC/3.0.0.0 MediationServer ALLOW: Ack, Cancel, Bye,Invite v=0 o=- 0 0 IN IP4 192.168.51.16 s=session c=IN IP4 192.168.51.16 b=CT:1000 t=0 0 m=audio 60372 RTP/AVP 0 101 c=IN IP4 192.168.51.16 a=rtcp:60373 a=label:Audio a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (13 headers 14 lines) --- csgtacsip1*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.51.16:60372 Got unsupported a:rtcp in SDP offer Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.51.16:60372 list_route: hop: csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.16, port 5060 csgtacsip1*CLI> Transmitting (no NAT) to 192.168.51.16:5060: ACK sip:csgredocm.csgopenline.com:5060;transport=Tcp;maddr=192.168.51.16 SIP/2.0 Via: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK2a8a0931;rport Max-Forwards: 70 From: "citywok" ;tag=as23a917d8 To: ;tag=c296554458 Contact: Call-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.13 Content-Length: 0 --- csgtacsip1*CLI> -- SIP/OCS_TRUNK-f7628128 answered SIP/1593-08f5a370 csgtacsip1*CLI> Audio is at 192.168.51.15 port 10420 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKa9a8472a2;received=10.0.11.1 From: citywok SVR1 ;tag=5230473a2e5a566 To: citywok SVR1 ;tag=as268797a3 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797646 INVITE User-Agent: Asterisk PBX 1.6.0.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 1065674434 1065674434 IN IP4 192.168.51.15 s=Asterisk PBX 1.6.0.13 c=IN IP4 192.168.51.15 t=0 0 m=audio 10420 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/1593-08f5a370 and SIP/OCS_TRUNK-f7628128 csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> ACK sip:1593@192.168.51.15 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK8cbccbc7c Max-Forwards: 70 Content-Length: 0 To: citywok SVR1 ;tag=as268797a3 From: citywok SVR1 ;tag=5230473a2e5a566 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797646 ACK Contact: citywok SVR1 Authorization:Digest response="6f6eabe0abe2b90e2ec3aae089d35d80",username="1593",realm="asterisk",nonce="471d56f1",algorithm=MD5,uri="sip:1593@192.168.51.15:5060" User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> BYE sip:1593@192.168.51.15 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK81da7335b Max-Forwards: 70 Content-Length: 0 To: citywok SVR1 ;tag=as268797a3 From: citywok SVR1 ;tag=5230473a2e5a566 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797647 BYE Supported: timer Supported: replaces Authorization:Digest response="2ebd706b2bfaef6f666eed2098bb52ec",username="1593",realm="asterisk",nonce="471d56f1",algorithm=MD5,uri="sip:1593@192.168.51.15" User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (12 headers 0 lines) --- csgtacsip1*CLI> Sending to 10.0.11.1 : 1221 (NAT) csgtacsip1*CLI> <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK81da7335b;received=10.0.11.1 From: citywok SVR1 ;tag=5230473a2e5a566 To: citywok SVR1 ;tag=as268797a3 Call-ID: 33e7c9839c51a59b871b2157e53c4c69@192.168.16.221 CSeq: 2048797647 BYE User-Agent: Asterisk PBX 1.6.0.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> csgtacsip1*CLI> Scheduling destruction of SIP dialog '1e4737952034f490361f3f581b868734@192.168.51.15' in 32000 ms (Method: INVITE) csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.16, port 5060 csgtacsip1*CLI> Reliably Transmitting (no NAT) to 192.168.51.16:5060: BYE sip:csgredocm.csgopenline.com:5060;transport=Tcp;maddr=192.168.51.16 SIP/2.0 Via: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK5a0db1c2;rport Max-Forwards: 70 From: "citywok" ;tag=as23a917d8 To: ;tag=c296554458 Call-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.13 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 csgtacsip1*CLI> Content-Length: 0 --- == Spawn extension (from-staff, 1593, 1) exited non-zero on 'SIP/1593-08f5a370' csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 200 OK FROM: "citywok";tag=as23a917d8 TO: ;tag=c296554458;epid=FCE3547589 CSEQ: 103 BYE CALL-ID: 1e4737952034f490361f3f581b868734@192.168.51.15 VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK5a0db1c2;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> Disconnected from Asterisk server csgtacsip1:~/asterisk-1.6.0.13#