root@asterisk:~# asterisk -R Asterisk 1.6.2.0-rc6, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details. ========================================================================= [Nov 18 16:37:33] Connected to Asterisk 1.6.2.0-rc6 currently running on asterisk (pid = 9311) Verbosity is at least 50 Core debug is at least 50 [2009-11-18 16:37:35] <--- SIP read from UDP:192.168.0.125:64877 ---> INVITE sip:18479227343@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjzE9-4fD0sJSLHWFxrgyXjmvAscQ3JQSQ Max-Forwards: 70 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: Contact: "Brendan Martens" Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24228 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Telephone 0.14.3 Content-Type: application/sdp Content-Length: 462 v=0 o=- 3467569055 3467569055 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 a=X-nat:0 m=audio 4012 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4013 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-11-18 16:37:35] --- (13 headers 20 lines) --- [2009-11-18 16:37:35] == Using SIP RTP TOS bits 16 [2009-11-18 16:37:35] == Using SIP RTP CoS mark 5 [2009-11-18 16:37:35] == Using UDPTL TOS bits 16 [2009-11-18 16:37:35] == Using UDPTL CoS mark 5 [2009-11-18 16:37:35] Sending to 192.168.0.125 : 64877 (no NAT) [2009-11-18 16:37:35] Using INVITE request as basis request - vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 [2009-11-18 16:37:35] Found peer 'brendanmartens' for 'brendanmartens' from 192.168.0.125:64877 [2009-11-18 16:37:35] <--- Reliably Transmitting (NAT) to 192.168.0.125:64877 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjzE9-4fD0sJSLHWFxrgyXjmvAscQ3JQSQ;received=192.168.0.125 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as4eee1c3c Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24228 INVITE Server: Asterisk PBX 1.6.2.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="59abd311" Content-Length: 0 <------------> [2009-11-18 16:37:35] Scheduling destruction of SIP dialog 'vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8' in 32000 ms (Method: INVITE) [2009-11-18 16:37:35] <--- SIP read from UDP:192.168.0.125:64877 ---> ACK sip:18479227343@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjzE9-4fD0sJSLHWFxrgyXjmvAscQ3JQSQ Max-Forwards: 70 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as4eee1c3c Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24228 ACK Content-Length: 0 <-------------> [2009-11-18 16:37:35] --- (8 headers 0 lines) --- [2009-11-18 16:37:35] <--- SIP read from UDP:192.168.0.125:64877 ---> INVITE sip:18479227343@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjktEdwyKqzypLNUTw-u61Opqpvs02Vvee Max-Forwards: 70 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: Contact: "Brendan Martens" Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24229 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Telephone 0.14.3 Authorization: Digest username="brendanmartens", realm="asterisk.crosscomm.net", nonce="59abd311", uri="sip:18479227343@asterisk.crosscomm.net", response="307b99bdb7ed618cb77f8d259aa307d7", algorithm=MD5 Content-Type: application/sdp Content-Length: 462 v=0 o=- 3467569055 3467569055 IN IP4 192.168.0.125 s=pjmedia c=IN IP4 192.168.0.125 t=0 0 a=X-nat:0 m=audio 4012 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4013 IN IP4 192.168.0.125 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-11-18 16:37:35] --- (14 headers 20 lines) --- [2009-11-18 16:37:35] Sending to 192.168.0.125 : 64877 (NAT) [2009-11-18 16:37:35] Using INVITE request as basis request - vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 [2009-11-18 16:37:35] Found peer 'brendanmartens' for 'brendanmartens' from 192.168.0.125:64877 [2009-11-18 16:37:35] Found RTP audio format 103 [2009-11-18 16:37:35] Found RTP audio format 102 [2009-11-18 16:37:35] Found RTP audio format 104 [2009-11-18 16:37:35] Found RTP audio format 117 [2009-11-18 16:37:35] Found RTP audio format 3 [2009-11-18 16:37:35] Found RTP audio format 0 [2009-11-18 16:37:35] Found RTP audio format 8 [2009-11-18 16:37:35] Found RTP audio format 9 [2009-11-18 16:37:35] Found RTP audio format 101 [2009-11-18 16:37:35] Found audio description format speex for ID 103 [2009-11-18 16:37:35] Found audio description format speex for ID 102 [2009-11-18 16:37:35] Found audio description format speex for ID 104 [2009-11-18 16:37:35] Found audio description format iLBC for ID 117 [2009-11-18 16:37:35] Found audio description format GSM for ID 3 [2009-11-18 16:37:35] Found audio description format PCMU for ID 0 [2009-11-18 16:37:35] Found audio description format PCMA for ID 8 [2009-11-18 16:37:35] Found audio description format G722 for ID 9 [2009-11-18 16:37:35] Found audio description format telephone-event for ID 101 [2009-11-18 16:37:35] Capabilities: us - 0x1fdf (g723|gsm|ulaw|alaw|g726|slin|lpc10|g729|speex|ilbc|g726aal2|g722), peer - audio=0x50160e (gsm|ulaw|alaw|speex|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x160e (gsm|ulaw|alaw|speex|ilbc|g722) [2009-11-18 16:37:35] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-11-18 16:37:35] Peer audio RTP is at port 192.168.0.125:4012 [2009-11-18 16:37:35] Looking for 18479227343 in softphones (domain asterisk.crosscomm.net) [2009-11-18 16:37:35] list_route: hop: [2009-11-18 16:37:35] <--- Transmitting (NAT) to 192.168.0.125:64877 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjktEdwyKqzypLNUTw-u61Opqpvs02Vvee;received=192.168.0.125 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24229 INVITE Server: Asterisk PBX 1.6.2.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-11-18 16:37:35] -- Executing [18479227343@softphones:1] Macro("SIP/brendanmartens-00000006", "outgoing,18479227343,9196679432,CrossComm") in new stack [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:1] Set("SIP/brendanmartens-00000006", "CALLERID(num)=9196679432") in new stack [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:2] Set("SIP/brendanmartens-00000006", "CALLERID(name)=CrossComm") in new stack [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:3] NoOp("SIP/brendanmartens-00000006", "SIPCALLID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8") in new stack [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:4] NoOp("SIP/brendanmartens-00000006", "Calling with Flowroute") in new stack [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:5] Answer("SIP/brendanmartens-00000006", "") in new stack [2009-11-18 16:37:35] Audio is at 98.101.39.108 port 20014 [2009-11-18 16:37:35] Adding codec 0x2 (gsm) to SDP [2009-11-18 16:37:35] Adding codec 0x4 (ulaw) to SDP [2009-11-18 16:37:35] Adding codec 0x8 (alaw) to SDP [2009-11-18 16:37:35] Adding codec 0x200 (speex) to SDP [2009-11-18 16:37:35] Adding codec 0x400 (ilbc) to SDP [2009-11-18 16:37:35] Adding codec 0x1000 (g722) to SDP [2009-11-18 16:37:35] Adding non-codec 0x1 (telephone-event) to SDP [2009-11-18 16:37:35] <--- Reliably Transmitting (NAT) to 192.168.0.125:64877 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjktEdwyKqzypLNUTw-u61Opqpvs02Vvee;received=192.168.0.125 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as518b9bdc Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24229 INVITE Server: Asterisk PBX 1.6.2.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 417 v=0 o=root 1987121872 1987121872 IN IP4 98.101.39.108 s=Asterisk PBX 1.6.2.0-rc6 c=IN IP4 98.101.39.108 t=0 0 m=audio 20014 RTP/AVP 3 0 8 102 117 9 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 speex/8000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-11-18 16:37:35] <--- SIP read from UDP:192.168.0.125:64877 ---> ACK sip:18479227343@98.101.39.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjZ4XLj3kT4UKs5mz9KB2yoYqW1awDOhgi Max-Forwards: 70 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as518b9bdc Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24229 ACK Content-Length: 0 <-------------> [2009-11-18 16:37:35] --- (8 headers 0 lines) --- [2009-11-18 16:37:35] -- Executing [s@macro-outgoing:6] Dial("SIP/brendanmartens-00000006", "SIP/18479227343@flowroute,,kKtT") in new stack [2009-11-18 16:37:35] == Using SIP RTP TOS bits 16 [2009-11-18 16:37:35] == Using SIP RTP CoS mark 5 [2009-11-18 16:37:35] == Using UDPTL TOS bits 16 [2009-11-18 16:37:35] == Using UDPTL CoS mark 5 [2009-11-18 16:37:35] Audio is at 98.101.39.108 port 20016 [2009-11-18 16:37:35] Adding codec 0x2 (gsm) to SDP [2009-11-18 16:37:35] Adding codec 0x4 (ulaw) to SDP [2009-11-18 16:37:35] Adding codec 0x8 (alaw) to SDP [2009-11-18 16:37:35] Adding non-codec 0x1 (telephone-event) to SDP [2009-11-18 16:37:35] Reliably Transmitting (no NAT) to 70.167.153.130:5060: INVITE sip:18479227343@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7d58ebc2;rport Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: Contact: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0-rc6 Date: Wed, 18 Nov 2009 21:37:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 316 v=0 o=root 1847455978 1847455978 IN IP4 98.101.39.108 s=Asterisk PBX 1.6.2.0-rc6 c=IN IP4 98.101.39.108 t=0 0 m=audio 20016 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-11-18 16:37:35] -- Called 18479227343@flowroute [2009-11-18 16:37:35] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7d58ebc2;rport=5060 From: "CrossComm" ;tag=as127c1ea9 To: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 102 INVITE Content-Length: 0 <-------------> [2009-11-18 16:37:35] --- (7 headers 0 lines) --- [2009-11-18 16:37:35] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7d58ebc2;rport=5060 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=e9901e48385c1aec25ff857f21344e5a.149a Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="4b04693d00010d6ebc003fdf4b164d8eb23b088121f69404", qop="auth" Content-Length: 0 <-------------> 2009-11-18 16:37:35] --- (8 headers 0 lines) --- [2009-11-18 16:37:35] Transmitting (no NAT) to 70.167.153.130:5060: ACK sip:18479227343@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7d58ebc2;rport Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=e9901e48385c1aec25ff857f21344e5a.149a Contact: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.0-rc6 Content-Length: 0 --- [2009-11-18 16:37:35] Audio is at 98.101.39.108 port 20016 [2009-11-18 16:37:35] Adding codec 0x2 (gsm) to SDP [2009-11-18 16:37:35] Adding codec 0x4 (ulaw) to SDP [2009-11-18 16:37:35] Adding codec 0x8 (alaw) to SDP [2009-11-18 16:37:35] Adding non-codec 0x1 (telephone-event) to SDP [2009-11-18 16:37:35] Reliably Transmitting (no NAT) to 70.167.153.130:5060: INVITE sip:18479227343@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK33804a00;rport Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: Contact: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.0-rc6 Proxy-Authorization: Digest username="57430558", realm="sip.flowroute.com", algorithm=MD5, uri="sip:18479227343@sip.flowroute.com", nonce="4b04693d00010d6ebc003fdf4b164d8eb23b088121f69404", response="0e110c790848c219ead584042ce2a4ae", qop=auth, cnonce="367f473f", nc=00000001 Date: Wed, 18 Nov 2009 21:37:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 316 v=0 o=root 1847455978 1847455979 IN IP4 98.101.39.108 s=Asterisk PBX 1.6.2.0-rc6 c=IN IP4 98.101.39.108 t=0 0 m=audio 20016 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-11-18 16:37:35] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK33804a00;rport=5060 From: "CrossComm" ;tag=as127c1ea9 To: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 103 INVITE Content-Length: 0 <-------------> [2009-11-18 16:37:35] --- (7 headers 0 lines) --- [2009-11-18 16:37:36] <--- SIP read from UDP:192.168.0.125:64877 ---> <-------------> [2009-11-18 16:37:37] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 98.101.39.108:5060;received=98.101.39.108;branch=z9hG4bK33804a00;rport=5060 From: ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 103 INVITE Content-Type: application/sdp Contact: Supported: timer,100rel Content-Length: 230 Record-Route: Record-Route: v=0 o=Flowroute.com 16122 26203 IN IP4 64.194.139.61 s=Flowroute.com c=IN IP4 64.194.139.61 t=0 0 m=audio 33124 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-------------> [2009-11-18 16:37:37] --- (12 headers 11 lines) --- [2009-11-18 16:37:37] Found RTP audio format 0 [2009-11-18 16:37:37] Found RTP audio format 101 [2009-11-18 16:37:37] Found audio description format PCMU for ID 0 [2009-11-18 16:37:37] Found audio description format telephone-event for ID 101 [2009-11-18 16:37:37] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-11-18 16:37:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-11-18 16:37:37] Peer audio RTP is at port 64.194.139.61:33124 [2009-11-18 16:37:37] -- SIP/flowroute-00000007 is making progress passing it to SIP/brendanmartens-00000006 [2009-11-18 16:37:41] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 98.101.39.108:5060;received=98.101.39.108;branch=z9hG4bK33804a00;rport=5060 From: ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 103 INVITE Content-Type: application/sdp Contact: Supported: timer,100rel Content-Length: 230 Record-Route: Record-Route: v=0 o=Flowroute.com 16122 26203 IN IP4 64.194.139.61 s=Flowroute.com c=IN IP4 64.194.139.61 t=0 0 m=audio 33124 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-------------> [2009-11-18 16:37:41] --- (12 headers 11 lines) --- [2009-11-18 16:37:41] list_route: hop: [2009-11-18 16:37:41] list_route: hop: [2009-11-18 16:37:41] set_destination: Parsing for address/port to send to [2009-11-18 16:37:41] set_destination: set destination to 70.167.153.130, port 5060 [2009-11-18 16:37:41] Transmitting (no NAT) to 70.167.153.130:5060: ACK sip:18479227343@64.194.139.61:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK32168e71;rport Route: , Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=30178 Contact: Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.0-rc6 Content-Length: 0 --- [2009-11-18 16:37:41] set_destination: Parsing for address/port to send to [2009-11-18 16:37:41] set_destination: set destination to 70.167.153.130, port 5060 [2009-11-18 16:37:41] Reliably Transmitting (no NAT) to 70.167.153.130:5060: BYE sip:18479227343@64.194.139.61:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7f65c691;rport Route: , Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 104 BYE User-Agent: Asterisk PBX 1.6.2.0-rc6 Proxy-Authorization: Digest username="57430558", realm="sip.flowroute.com", algorithm=MD5, uri="sip:18479227343@64.194.139.61:5060", nonce="4b04693d00010d6ebc003fdf4b164d8eb23b088121f69404", response="a1faeebb87b69812d9aabe3241df38d3", qop=auth, cnonce="0d091eab", nc=00000002 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [2009-11-18 16:37:41] Scheduling destruction of SIP dialog '53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com' in 32000 ms (Method: INVITE) [2009-11-18 16:37:41] -- SIP/flowroute-00000007 answered SIP/brendanmartens-00000006 [2009-11-18 16:37:41] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK7f65c691;rport=5060 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 104 BYE Content-Length: 0 <-------------> [2009-11-18 16:37:41] --- (7 headers 0 lines) --- [2009-11-18 16:37:41] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 98.101.39.108:5060;received=98.101.39.108;branch=z9hG4bK7f65c691;rport=5060 From: ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 104 BYE Contact: Supported: timer,100rel Content-Length: 0 <-------------> [2009-11-18 16:37:41] --- (9 headers 0 lines) --- [2009-11-18 16:37:44] <--- SIP read from UDP:192.168.0.125:64877 ---> BYE sip:18479227343@98.101.39.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjbMokZS1PWJooBp0RC18iuKcaKuWCO-04 Max-Forwards: 70 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as518b9bdc Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24230 BYE User-Agent: Telephone 0.14.3 Content-Length: 0 <-------------> [2009-11-18 16:37:44] --- (9 headers 0 lines) --- [2009-11-18 16:37:44] Sending to 192.168.0.125 : 64877 (NAT) [2009-11-18 16:37:44] <--- Transmitting (NAT) to 192.168.0.125:64877 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.125:64877;rport;branch=z9hG4bKPjbMokZS1PWJooBp0RC18iuKcaKuWCO-04;received=192.168.0.125 From: "Brendan Martens" ;tag=nT.8AnF3iMMm5dO8oRoG6gkMfpZtWozq To: ;tag=as518b9bdc Call-ID: vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8 CSeq: 24230 BYE Server: Asterisk PBX 1.6.2.0-rc6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-11-18 16:37:44] Scheduling destruction of SIP dialog '53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com' in 32000 ms (Method: INVITE) [2009-11-18 16:37:44] set_destination: Parsing for address/port to send to [2009-11-18 16:37:44] set_destination: set destination to 70.167.153.130, port 5060 [2009-11-18 16:37:44] Reliably Transmitting (no NAT) to 70.167.153.130:5060: BYE sip:18479227343@64.194.139.61:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK1e0d30e3;rport Route: , Max-Forwards: 70 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 105 BYE User-Agent: Asterisk PBX 1.6.2.0-rc6 Proxy-Authorization: Digest username="57430558", realm="sip.flowroute.com", algorithm=MD5, uri="sip:18479227343@64.194.139.61:5060", nonce="4b04693d00010d6ebc003fdf4b164d8eb23b088121f69404", response="30f932e846fcc6792a35b43ed342da05", qop=auth, cnonce="5470bf78", nc=00000003 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2009-11-18 16:37:44] == Spawn extension (macro-outgoing, s, 6) exited non-zero on 'SIP/brendanmartens-00000006' in macro 'outgoing' [2009-11-18 16:37:44] == Spawn extension (softphones, 18479227343, 1) exited non-zero on 'SIP/brendanmartens-00000006' [2009-11-18 16:37:44] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 98.101.39.108:5060;branch=z9hG4bK1e0d30e3;rport=5060 From: "CrossComm" ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 105 BYE Content-Length: 0 <-------------> [2009-11-18 16:37:44] --- (7 headers 0 lines) --- [2009-11-18 16:37:44] Really destroying SIP dialog 'vEaJY0Gp.F.QoJp-zJ7wasfj70d3xQv8' Method: BYE [2009-11-18 16:37:44] <--- SIP read from UDP:70.167.153.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 98.101.39.108:5060;received=98.101.39.108;branch=z9hG4bK1e0d30e3;rport=5060 From: ;tag=as127c1ea9 To: ;tag=30178 Call-ID: 53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com CSeq: 105 BYE Contact: Supported: timer,100rel Content-Length: 0 <-------------> [2009-11-18 16:37:44] --- (9 headers 0 lines) --- [2009-11-18 16:37:44] Really destroying SIP dialog '53c4b5ad047af8342bd6ae3538da99c3@sip.flowroute.com' Method: INVITE