Asterisk 1.2.17, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.17 currently running on fargo (pid = 14789) fargo*CLI> Verbosity is at least 4 Core debug is at least 4 fargo*CLI> Destroying call 'fa3c2d726d978f01' fargo*CLI> <-- SIP read from 10.50.103.51:1036: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKec0ba85d5f72adfe5 Max-Forwards: 70 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 Call-ID: 55227f33c746ac3c CSeq: 24543 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt User-Agent: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (13 headers 24 lines) --- Using INVITE request as basis request - 55227f33c746ac3c Sending to 10.50.103.51 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKec0ba85d5f72adfe5;received=10.50.103.51 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 ;tag=as17652136 Call-ID: 55227f33c746ac3c CSeq: 24543 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5134ff1f" Content-Length: 0 --- Scheduling destruction of call '55227f33c746ac3c' in 15000 ms Found user '42511' fargo*CLI> <-- SIP read from 10.50.103.51:1036: ACK sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKec0ba85d5f72adfe5 Max-Forwards: 70 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 ;tag=as17652136 Call-ID: 55227f33c746ac3c CSeq: 24543 ACK User-Agent: Aastra 55i/2.1.0.2067 Content-Length: 0 --- (9 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.51:1036: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKcf19628318390ee26 Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="5134ff1f",uri="sip:42512@fargo.ana.aastra.com:6050",response="55f4d44500832b7e530938c84ac4b97c",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 Call-ID: 55227f33c746ac3c CSeq: 24544 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt User-Agent: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 24 lines) --- Using INVITE request as basis request - 55227f33c746ac3c Sending to 10.50.103.51 : 5060 (non-NAT) Found user '42511' fargo*CLI> Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 42512 in default (domain fargo.ana.aastra.com) list_route: hop: Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKcf19628318390ee26;received=10.50.103.51 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 Call-ID: 55227f33c746ac3c CSeq: 24544 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> -- Executing Dial("SIP/42511-081e5aa8", "SIP/42512") in new stack We're at 10.50.103.10 port 17020 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.50.103.54:5060: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6ce8e5c2;rport From: "Robert Arritt" ;tag=as54164a74 To: Contact: Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:40:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 14789 14789 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 17020 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Reliably Transmitting (no NAT) to 10.50.103.54:5060: NOTIFY sip:42512@10.50.103.54:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK41e8bd4f;rport From: ;tag=as63ece84f To: Robert Arritt ;tag=ca6ac91818 Contact: Call-ID: 59fc18cfff02764e CSeq: 115 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 220 confirmed --- Extension Changed 42511 new state InUse for Notify User 42512 fargo*CLI> -- Called 42512 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6ce8e5c2;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Server: Aastra 35i/20070327 Content-Length: 0 --- (11 headers 0 lines) --- fargo*CLI> -- SIP/42512-081d3720 is ringing Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKcf19628318390ee26;received=10.50.103.51 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 ;tag=as71278b5f Call-ID: 55227f33c746ac3c CSeq: 24544 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK41e8bd4f;rport=6050;received=10.50.103.10 From: ;tag=as63ece84f To: Robert Arritt ;tag=ca6ac91818 Call-ID: 59fc18cfff02764e CSeq: 115 NOTIFY Contact: Robert Arritt Server: Aastra 35i/20070327 Content-Length: 0 --- (9 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6ce8e5c2;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 102 INVITE Contact: Robert Arritt Server: Aastra 35i/20070327 Supported: timer Content-Type: application/sdp Content-Length: 256 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (11 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Transmitting (no NAT) to 10.50.103.54:5060: ACK sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK57846df0;rport From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Contact: Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> -- SIP/42512-081d3720 answered SIP/42511-081e5aa8 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKcf19628318390ee26;received=10.50.103.51 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 ;tag=as71278b5f Call-ID: 55227f33c746ac3c CSeq: 24544 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 14789 14789 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 15946 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/42511-081e5aa8 and SIP/42512-081d3720 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 We're at 10.50.103.10 port 17020 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (no NAT) to 10.50.103.54:5060: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3edf8a07;rport From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Contact: Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 321 v=0 o=root 14789 14790 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> <-- SIP read from 10.50.103.51:1036: ACK sip:42512@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK3015e89ca475d3b6e Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="5134ff1f",uri="sip:42512@10.50.103.10:6050",response="30b0f8304c44f7b189be6d14aaa68b51",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=c8c7e3d392 To: 42512 ;tag=as71278b5f Call-ID: 55227f33c746ac3c CSeq: 24544 ACK User-Agent: Aastra 55i/2.1.0.2067 Content-Length: 0 --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK76516ae2;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 14789 14790 IN IP4 10.50.103.54 s=session c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3edf8a07;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 103 INVITE Server: Aastra 35i/20070327 Content-Length: 0 --- (8 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3edf8a07;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 103 INVITE Contact: Robert Arritt Server: Aastra 35i/20070327 Supported: timer Content-Type: application/sdp Content-Length: 315 v=0 o=MxSIP 0 3664 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (11 headers 15 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 10 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format L16 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x5c (ulaw|alaw|g726|slin)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Transmitting (no NAT) to 10.50.103.54:5060: ACK sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6aa24694;rport From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Contact: Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK76516ae2;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 102 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 26640 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK43c5f925;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2d62633a;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 295 v=0 o=root 14789 14791 IN IP4 10.50.103.54 s=session c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2d62633a;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 103 INVITE Server: Aastra 55i/2.1.0.2067 Content-Length: 0 --- (8 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2d62633a;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 103 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 316 v=0 o=MxSIP 0 31761 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 15 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 10 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format L16 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x5c (ulaw|alaw|g726|slin)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7a2fd3a8;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK4b14a9283b4a946db Max-Forwards: 70 From: ;tag=3490754866 To: "Robert Arritt" ;tag=as54164a74 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 8102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 564 v=0 o=MxSIP 0 1 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 99 10 98 97 111 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:99 PCMA/16000 a=rtpmap:10 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendonly --- (14 headers 24 lines) --- Using INVITE request as basis request - 19481c5b753760013c1924b37253eab2@10.50.103.10 Sending to 10.50.103.54 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 99 Found RTP audio format 10 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 111 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 fargo*CLI> Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on channel 'SIP/42511-081e5aa8' We're at 10.50.103.10 port 17020 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK4b14a9283b4a946db;received=10.50.103.54 From: ;tag=3490754866 To: "Robert Arritt" ;tag=as54164a74 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 8102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 321 v=0 o=root 14789 14791 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0b35c3bb;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 14789 14792 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 fargo*CLI> m=audio 15946 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> <-- SIP read from 10.50.103.54:1036: ACK sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK502bdfdba47323885 Max-Forwards: 70 From: ;tag=3490754866 To: "Robert Arritt" ;tag=as54164a74 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 8102 ACK User-Agent: Aastra 35i/20070327 Content-Length: 0 --- (9 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0b35c3bb;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 104 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 23707 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK56f3f30a;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> Mar 27 11:41:05 NOTICE[14793]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! fargo*CLI> Destroying call 'fbc114da70e7bcc173259740def5f93f@10.50.103.52' fargo*CLI> -- Stopped music on hold on SIP/42511-081e5aa8 fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK330e558c4d63c86a4 Max-Forwards: 70 From: Robert Arritt ;tag=74d8fdd51e To: 42514 Call-ID: d54abf341d5ed84b CSeq: 9712 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 24 lines) --- Using INVITE request as basis request - d54abf341d5ed84b Sending to 10.50.103.54 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK330e558c4d63c86a4;received=10.50.103.54 From: Robert Arritt ;tag=74d8fdd51e To: 42514 ;tag=as1ee638eb Call-ID: d54abf341d5ed84b CSeq: 9712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7da13a45" Content-Length: 0 --- Scheduling destruction of call 'd54abf341d5ed84b' in 15000 ms Found user '42512' fargo*CLI> <-- SIP read from 10.50.103.54:1036: ACK sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK330e558c4d63c86a4 Max-Forwards: 70 From: Robert Arritt ;tag=74d8fdd51e To: 42514 ;tag=as1ee638eb Call-ID: d54abf341d5ed84b CSeq: 9712 ACK User-Agent: Aastra 35i/20070327 Content-Length: 0 --- (9 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKddc1dac931155c629 Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="7da13a45",uri="sip:42514@fargo.ana.aastra.com:6050",response="c37dcd06feb09b01df00e4def9703632",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=74d8fdd51e To: 42514 Call-ID: d54abf341d5ed84b CSeq: 9713 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 fargo*CLI> a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 24 lines) --- Using INVITE request as basis request - d54abf341d5ed84b Sending to 10.50.103.54 : 5060 (non-NAT) Found user '42512' Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3002 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) fargo*CLI> Looking for 42514 in default (domain fargo.ana.aastra.com) list_route: hop: Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKddc1dac931155c629;received=10.50.103.54 From: Robert Arritt ;tag=74d8fdd51e To: 42514 Call-ID: d54abf341d5ed84b CSeq: 9713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/42512-081d9310", "SIP/42514") in new stack We're at 10.50.103.10 port 16406 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport From: "Robert Arritt" ;tag=as282777fd To: Contact: Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:41:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 14789 14789 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 16406 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 42514 fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as282777fd To: ;tag=cbd7be2c1ea1fc9 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (8 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 180 Ringing Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as282777fd To: ;tag=cbd7be2c1ea1fc9 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport Content-Length: 0 Call-Info: ;appearance-index=1 Allow-Events: talk, hold, conference Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (11 headers 0 lines) --- -- SIP/42514-081f0308 is ringing Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKddc1dac931155c629;received=10.50.103.54 From: Robert Arritt ;tag=74d8fdd51e To: 42514 ;tag=as2eab4731 Call-ID: d54abf341d5ed84b CSeq: 9713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.54:1036: REFER sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKfc05e8084eadf385b Max-Forwards: 70 From: ;tag=3490754866 To: "Robert Arritt" ;tag=as54164a74 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 8103 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Refer-To: 42514 Supported: timer User-Agent: Aastra 35i/20070327 Content-Length: 0 --- (14 headers 0 lines) --- Mar 27 11:41:09 WARNING[14805]: chan_sip.c:6936 get_refer_info: Referred-by: Huh? Not a SIP header () Ignoring? Transfer to 42514 in default Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKfc05e8084eadf385b;received=10.50.103.54 From: ;tag=3490754866 To: "Robert Arritt" ;tag=as54164a74 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 8103 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Reliably Transmitting (no NAT) to 10.50.103.54:5060: NOTIFY sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK29e7afae;rport From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Contact: Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=8103 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Reliably Transmitting (no NAT) to 10.50.103.54:5060: BYE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK546e08ff;rport From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- == Spawn extension (default, 42514, 0) exited non-zero on 'SIP/42511-081e5aa8' -- Executing Dial("SIP/42511-081e5aa8", "SIP/42514") in new stack We're at 10.50.103.10 port 14932 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4f9ac98d;rport From: "Robert Arritt" ;tag=as1a824fa0 To: Contact: Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:41:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 14789 14789 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 14932 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 42514 fargo*CLI> <-- SIP read from 10.50.103.54:1036: BYE sip:42514@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKa32dbde3608e4d07c Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="7da13a45",uri="sip:42514@10.50.103.10:6050",response="192148757f7b2eb31a5b1283b2cf7a73",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=74d8fdd51e To: 42514 ;tag=as2eab4731 Call-ID: d54abf341d5ed84b CSeq: 9714 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Supported: timer User-Agent: Aastra 35i/20070327 Content-Length: 0 --- (13 headers 0 lines) --- Sending to 10.50.103.54 : 5060 (non-NAT) Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKa32dbde3608e4d07c;received=10.50.103.54 From: Robert Arritt ;tag=74d8fdd51e To: 42514 ;tag=as2eab4731 Call-ID: d54abf341d5ed84b CSeq: 9714 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> Scheduling destruction of call '032df1b9002e9a255516060622c8b594@10.50.103.10' in 32000 ms Reliably Transmitting (no NAT) to 10.50.103.52:5060: CANCEL sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport From: "Robert Arritt" ;tag=as282777fd To: Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 42514, 1) exited non-zero on 'SIP/42512-081d9310' Mar 27 11:41:09 ERROR[20648]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 400 Missing status-line Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK29e7afae;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 104 NOTIFY Server: Aastra 35i/20070327 Content-Length: 0 --- (8 headers 0 lines) --- -- Incoming call: Got SIP response 400 "Missing status-line" back from 10.50.103.54 fargo*CLI> Destroying call 'd54abf341d5ed84b' fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK546e08ff;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as54164a74 To: ;tag=3490754866 Call-ID: 19481c5b753760013c1924b37253eab2@10.50.103.10 CSeq: 105 BYE Server: Aastra 35i/20070327 Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '19481c5b753760013c1924b37253eab2@10.50.103.10' fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 CANCEL From: "Robert Arritt" ;tag=as282777fd To: ;tag=cbd7be2c1ea1fc9 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport Content-Length: 0 Contact: 42514 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (10 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 487 Request Terminated Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as282777fd To: ;tag=cbd7be2c1ea1fc9 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (8 headers 0 lines) --- Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ab1aa2e;rport From: "Robert Arritt" ;tag=as282777fd To: ;tag=cbd7be2c1ea1fc9 Contact: Call-ID: 032df1b9002e9a255516060622c8b594@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4f9ac98d;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (8 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 180 Ringing Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4f9ac98d;rport Content-Length: 0 Call-Info: ;appearance-index=1 Allow-Events: talk, hold, conference Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (11 headers 0 lines) --- fargo*CLI> -- SIP/42514-081f5848 is ringing fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4f9ac98d;rport Content-Length: 242 Call-Info: ;appearance-index=1 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1026767575 IN IP4 10.50.103.52 s=SIP Call c=IN IP4 10.50.103.52 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - --- (14 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.52:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.52, port 5060 Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3a2d2886;rport From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Contact: Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> -- SIP/42514-081f5848 answered SIP/42511-081e5aa8 -- Attempting native bridge of SIP/42511-081e5aa8 and SIP/42514-081f5848 fargo*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK302c9f61;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 105 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 14789 14793 IN IP4 10.50.103.52 s=session c=IN IP4 10.50.103.52 t=0 0 fargo*CLI> m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.52, port 5060 We're at 10.50.103.10 port 14932 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK24638bef;rport From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Contact: Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 14789 14790 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK302c9f61;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 105 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 25526 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK424872bc;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 103 INVITE From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK24638bef;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (8 headers 0 lines) --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 103 INVITE From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK24638bef;rport Content-Length: 242 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1026767575 IN IP4 10.50.103.52 s=SIP Call c=IN IP4 10.50.103.52 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.52:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.52, port 5060 Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK796ed571;rport From: "Robert Arritt" ;tag=as1a824fa0 To: ;tag=b3818c3cbaee84f Contact: Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: BYE sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.52;branch=z9hG4bKce9ba9fba Max-Forwards: 70 Content-Length: 0 To: "Robert Arritt" ;tag=as1a824fa0 From: ;tag=b3818c3cbaee84f Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 145471998 BYE Supported: timer Call-Info: ;appearance-index=1 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 --- (12 headers 0 lines) --- Sending to 10.50.103.52 : 5060 (non-NAT) Transmitting (no NAT) to 10.50.103.52:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.52;branch=z9hG4bKce9ba9fba;received=10.50.103.52 From: ;tag=b3818c3cbaee84f To: "Robert Arritt" ;tag=as1a824fa0 Call-ID: 2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10 CSeq: 145471998 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- fargo*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 15946 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2be89263;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 106 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 14789 14794 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 fargo*CLI> m=audio 15946 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Spawn extension (default, 42514, 1) exited non-zero on 'SIP/42511-081e5aa8' Mar 27 11:41:12 ERROR[20642]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied Scheduling destruction of call '55227f33c746ac3c' in 32000 ms fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.54:5060: NOTIFY sip:42512@10.50.103.54:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ffea903;rport From: ;tag=as63ece84f To: Robert Arritt ;tag=ca6ac91818 Contact: Call-ID: 59fc18cfff02764e CSeq: 116 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 221 terminated --- Extension Changed 42511 new state Idle for Notify User 42512 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7ffea903;rport=6050;received=10.50.103.10 From: ;tag=as63ece84f To: Robert Arritt ;tag=ca6ac91818 Call-ID: 59fc18cfff02764e CSeq: 116 NOTIFY Contact: Robert Arritt Server: Aastra 35i/20070327 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '2c365a85439b7bbb1190ce6e20c80ca2@10.50.103.10' fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2be89263;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 106 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 32654 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK69520fad;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Contact: Call-ID: 55227f33c746ac3c CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Reliably Transmitting (no NAT) to 10.50.103.51:5060: BYE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0f6d2692;rport From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 107 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '55227f33c746ac3c' in 32000 ms fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0f6d2692;rport=6050;received=10.50.103.10 From: 42512 ;tag=as71278b5f To: Robert Arritt ;tag=c8c7e3d392 Call-ID: 55227f33c746ac3c CSeq: 107 BYE Server: Aastra 55i/2.1.0.2067 Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '55227f33c746ac3c' fargo*CLI> fargo*CLI>