Asterisk SVN-trunk-r81632M, Copyright (C) 1999 - 2007 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. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. Connected to Asterisk SVN-trunk-r81632M currently running on pbx-gr-backup (pid = 29375) pbx-gr-backup*CLI> Verbosity is at least 5 pbx-gr-backup*CLI> -- Remote UNIX connection pbx-gr-backup*CLI> core set verbose 5 pbx-gr-backup*CLI> Verbosity is at least 5 pbx-gr-backup*CLI> sip set debug on pbx-gr-backup*CLI> SIP Debugging enabled pbx-gr-backup*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Reliably Transmitting (NAT) to 132.64.4.137:2051: BYE sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK1f54906a;rport Max-Forwards: 70 From: "7106" ;tag=as25884f79 To: "Test" ;tag=cub1goodh4 Call-ID: 2c01fe540508962608fc75381fc1e87e@132.64.9.162 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- Scheduling destruction of SIP dialog '2c01fe540508962608fc75381fc1e87e@132.64.9.162' in 32000 ms (Method: BYE) pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK1f54906a;rport=5060 From: "7106" ;tag=as25884f79 To: "Test" ;tag=cub1goodh4 Call-ID: 2c01fe540508962608fc75381fc1e87e@132.64.9.162 CSeq: 104 BYE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '2c01fe540508962608fc75381fc1e87e@132.64.9.162' Method: BYE pbx-gr-backup*CLI> <--- SIP read from 128.139.26.6:57818 ---> INVITE sip:80609@132.64.9.162:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK7BF889 Remote-Party-ID: ;party=calling;screen=no;privacy=off From: ;tag=24257ABC-654 To: Date: Thu, 06 Sep 2007 05:29:17 GMT Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4065684744-1534071260-2234908698-801595720 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1189056557 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 4248 6846 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 18260 RTP/AVP 8 101 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (21 headers 11 lines) --- == Using TOS bits 0 == Using CoS mark 5 == Using TOS bits 0 == Using CoS mark 6 == Using TOS bits 0 == Using CoS mark 5 Sending to 128.139.26.6 : 5060 (no NAT) Using INVITE request as basis request - F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 pbx-gr-backup*CLI> No user '7106' in SIP users list pbx-gr-backup*CLI> Found peer '128.139.26.6' for '7106' from 128.139.26.6:57818 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 128.139.26.6:18260 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 128.139.26.6:18260 Looking for 80609 in huji-local (domain 132.64.9.162) list_route: hop: <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK7BF889;received=128.139.26.6 From: ;tag=24257ABC-654 To: Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> -- Executing [80609@huji-local:1] NoOp("SIP/gr-pbx-link-08222fe8", "") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:2] Set("SIP/gr-pbx-link-08222fe8", "_To=80609") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:3] Set("SIP/gr-pbx-link-08222fe8", "_From=7106") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:4] Set("SIP/gr-pbx-link-08222fe8", "DB(80609/LastCaller)=7106") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:5] Set("SIP/gr-pbx-link-08222fe8", "DB(7106/LastCalled)=80609") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:6] MYSQL("SIP/gr-pbx-link-08222fe8", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:7] MYSQL("SIP/gr-pbx-link-08222fe8", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='80609'") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:8] MYSQL("SIP/gr-pbx-link-08222fe8", "Fetch FetchId 2 aEXTEN") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:9] NoOp("SIP/gr-pbx-link-08222fe8", "") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:10] MYSQL("SIP/gr-pbx-link-08222fe8", "Clear 2") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:11] MYSQL("SIP/gr-pbx-link-08222fe8", "Query resID 1 SELECT callerid from sip_users where name='80609'") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:12] MYSQL("SIP/gr-pbx-link-08222fe8", "Fetch FetchId 2 CalledName") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:13] MYSQL("SIP/gr-pbx-link-08222fe8", "Clear 2") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:14] GotoIf("SIP/gr-pbx-link-08222fe8", "1?15:21") in new stack pbx-gr-backup*CLI> -- Goto (huji-local,80609,15) pbx-gr-backup*CLI> -- Executing [80609@huji-local:15] MYSQL("SIP/gr-pbx-link-08222fe8", "Query resID 1 SELECT name from full_names where number=7106") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:16] MYSQL("SIP/gr-pbx-link-08222fe8", "Fetch FetchId 2 CallingName") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:17] MYSQL("SIP/gr-pbx-link-08222fe8", "Clear 2") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:18] GotoIf("SIP/gr-pbx-link-08222fe8", "0?19:20") in new stack pbx-gr-backup*CLI> -- Goto (huji-local,80609,20) pbx-gr-backup*CLI> -- Executing [80609@huji-local:20] NoOp("SIP/gr-pbx-link-08222fe8", "Finish if-if-huji-local-4-5") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:21] NoOp("SIP/gr-pbx-link-08222fe8", "Finish if-huji-local-4") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:22] MYSQL("SIP/gr-pbx-link-08222fe8", "Disconnect 1") in new stack pbx-gr-backup*CLI> -- Executing [80609@huji-local:23] RetryDial("SIP/gr-pbx-link-08222fe8", ",10,5,SIP/80609,20,L(3600000:60000:30000)") in new stack pbx-gr-backup*CLI> -- Limit Data for this call: pbx-gr-backup*CLI> > timelimit = 3600000 pbx-gr-backup*CLI> > play_warning = 60000 pbx-gr-backup*CLI> > play_to_caller = yes pbx-gr-backup*CLI> > play_to_callee = no pbx-gr-backup*CLI> > warning_freq = 30000 pbx-gr-backup*CLI> > start_sound = pbx-gr-backup*CLI> > warning_sound = timeleft pbx-gr-backup*CLI> > end_sound = pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 5 pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 6 pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 5 pbx-gr-backup*CLI> Audio is at 132.64.9.162 port 41320 pbx-gr-backup*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr-backup*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr-backup*CLI> Reliably Transmitting (no NAT) to 132.64.4.137:2051: INVITE sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK140ba0f0;rport Max-Forwards: 70 From: "7106" ;tag=as34b7f090 To: Contact: Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Date: Thu, 06 Sep 2007 05:12:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 248 v=0 o=root 357543411 357543411 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 18260 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr-backup*CLI> -- Called 80609 pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK140ba0f0;rport=5060 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- pbx-gr-backup*CLI> -- SIP/80609-08226f78 is ringing <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK7BF889;received=128.139.26.6 From: ;tag=24257ABC-654 To: ;tag=as04f0a552 Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK140ba0f0;rport=5060 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- pbx-gr-backup*CLI> -- SIP/80609-08226f78 is ringing pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK140ba0f0;rport=5060 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1396818232 1396818233 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 49670 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/1ZefkyxHbGs/byZemPD85Wdp00+Wke9Mq3k5Vtx a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:49670 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:49670 --- set_address_from_contact host '132.64.4.137' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Transmitting (no NAT) to 132.64.4.137:2051: ACK sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6a824b77;rport Max-Forwards: 70 From: "7106" ;tag=as34b7f090 To: ;tag=bd1wdy40ux Contact: Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> -- SIP/80609-08226f78 answered SIP/gr-pbx-link-08222fe8 Audio is at 132.64.9.162 port 41562 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK7BF889;received=128.139.26.6 From: ;tag=24257ABC-654 To: ;tag=as04f0a552 Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 644232439 644232439 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 49670 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx-gr-backup*CLI> <--- SIP read from 128.139.26.6:57818 ---> ACK sip:80609@132.64.9.162:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK7C0E84 From: ;tag=24257ABC-654 To: ;tag=as04f0a552 Date: Thu, 06 Sep 2007 05:29:17 GMT Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK140ba0f0;rport=5060 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1396818232 1396818233 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 49670 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/1ZefkyxHbGs/byZemPD85Wdp00+Wke9Mq3k5Vtx a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:49670 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:49670 --- set_address_from_contact host '132.64.4.137' set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Transmitting (no NAT) to 132.64.4.137:2051: ACK sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK5b9f0b32;rport Max-Forwards: 70 From: "7106" ;tag=as34b7f090 To: ;tag=bd1wdy40ux Contact: Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> INVITE sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-6g9c5af3ald9;rport From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1396818232 1396818234 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 49670 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/1ZefkyxHbGs/byZemPD85Wdp00+Wke9Mq3k5Vtx a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendonly <-------------> --- (18 headers 19 lines) --- Sending to 132.64.4.137 : 2051 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:49670 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:49670 <--- Transmitting (NAT) to 132.64.4.137:2051 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-6g9c5af3ald9;received=132.64.4.137;rport=2051 From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 132.64.9.162 port 41320 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-6g9c5af3ald9;received=132.64.4.137;rport=2051 From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 357543411 357543412 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 18260 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> pbx-gr-backup*CLI> -- Started music on hold, class 'default', on SIP/gr-pbx-link-08222fe8 pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> ACK sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-821727fwxz2d;rport From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> INVITE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-58pysgwswzqt;rport From: "Test" ;tag=07wezivh5v To: Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 473 v=0 o=root 697328844 697328844 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 53624 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4ryPNGspKa1te9ajfSsXn/Q+rKeLGbLZ+0QVbi8q pbx-gr-backup*CLI> a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) --- == Using TOS bits 0 == Using CoS mark 5 == Using TOS bits 0 == Using CoS mark 6 == Using TOS bits 0 == Using CoS mark 5 Sending to 132.64.4.137 : 2051 (NAT) Using INVITE request as basis request - 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 Found user '80609' for '80609' <--- Reliably Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-58pysgwswzqt;received=132.64.4.137;rport=2051 From: "Test" ;tag=07wezivh5v To: ;tag=as3c22f8d2 Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="5446627b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4' in 32000 ms (Method: INVITE) pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-vy6bama0fg93;rport From: ;tag=2iswzd9g6i To: Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2051 <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-vy6bama0fg93;received=132.64.4.137;rport=2051 From: ;tag=2iswzd9g6i To: ;tag=as3eda8a50 Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="2c0285f2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> ACK sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-58pysgwswzqt;rport From: "Test" ;tag=07wezivh5v To: ;tag=as3c22f8d2 Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> INVITE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-iyte6i6dc4j0;rport From: "Test" ;tag=07wezivh5v To: Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="5446627b",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="f708a9be951e91a1010467b41430435e",algorithm=md5 Content-Type: application/sdp Content-Length: 473 v=0 o=root 697328844 697328844 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 53624 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4ryPNGspKa1te9ajfSsXn/Q+rKeLGbLZ+0QVbi8q a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 132.64.4.137 : 2051 (NAT) Using INVITE request as basis request - 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 Found user '80609' for '80609' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:53624 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0xc0008 (alaw|h261|h263), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:53624 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) list_route: hop: <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-iyte6i6dc4j0;received=132.64.4.137;rport=2051 From: "Test" ;tag=07wezivh5v To: Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:1] NoOp("SIP/80609-08280f60", "") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:2] Set("SIP/80609-08280f60", "_To=80620") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:3] Set("SIP/80609-08280f60", "_From=80609") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:4] Set("SIP/80609-08280f60", "DB(80620/LastCaller)=80609") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:5] Set("SIP/80609-08280f60", "DB(80609/LastCalled)=80620") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:6] MYSQL("SIP/80609-08280f60", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:7] MYSQL("SIP/80609-08280f60", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='80620'") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:8] MYSQL("SIP/80609-08280f60", "Fetch FetchId 2 aEXTEN") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:9] NoOp("SIP/80609-08280f60", "") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:10] MYSQL("SIP/80609-08280f60", "Clear 2") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:11] MYSQL("SIP/80609-08280f60", "Query resID 1 SELECT callerid from sip_users where name='80620'") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:12] MYSQL("SIP/80609-08280f60", "Fetch FetchId 2 CalledName") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:13] MYSQL("SIP/80609-08280f60", "Clear 2") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:14] GotoIf("SIP/80609-08280f60", "0?15:21") in new stack pbx-gr-backup*CLI> -- Goto (huji-remote-gr,80620,21) pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:21] NoOp("SIP/80609-08280f60", "Finish if-huji-local-4") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:22] MYSQL("SIP/80609-08280f60", "Disconnect 1") in new stack pbx-gr-backup*CLI> -- Executing [80620@huji-remote-gr:23] RetryDial("SIP/80609-08280f60", ",10,5,SIP/80620,20,L(3600000:60000:30000)") in new stack pbx-gr-backup*CLI> -- Limit Data for this call: pbx-gr-backup*CLI> > timelimit = 3600000 pbx-gr-backup*CLI> > play_warning = 60000 pbx-gr-backup*CLI> > play_to_caller = yes pbx-gr-backup*CLI> > play_to_callee = no pbx-gr-backup*CLI> > warning_freq = 30000 pbx-gr-backup*CLI> > start_sound = pbx-gr-backup*CLI> > warning_sound = timeleft pbx-gr-backup*CLI> > end_sound = pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 5 pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 6 pbx-gr-backup*CLI> == Using TOS bits 0 pbx-gr-backup*CLI> == Using CoS mark 5 pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-h76nx154cga6;rport From: ;tag=2iswzd9g6i To: Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="2c0285f2",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="57b8fb27e6fe700ff811b4ecb2d141f6",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2051 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) [Sep 6 08:12:51] NOTICE[29379]: chan_sip.c:15967 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-h76nx154cga6;received=132.64.4.137;rport=2051 From: ;tag=2iswzd9g6i To: ;tag=as3eda8a50 Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4' Method: SUBSCRIBE pbx-gr-backup*CLI> Audio is at 132.64.9.162 port 1920 pbx-gr-backup*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr-backup*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr-backup*CLI> Reliably Transmitting (no NAT) to 132.64.4.123:5060: INVITE sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e6dd32f;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as56538c2b To: Contact: Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Date: Thu, 06 Sep 2007 05:12:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 250 v=0 o=root 1336648888 1336648888 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 53624 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr-backup*CLI> -- Called 80620 pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 100 Trying Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e6dd32f;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (8 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 180 Ringing Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e6dd32f;rport Content-Length: 0 Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> pbx-gr-backup*CLI> --- (9 headers 0 lines) --- pbx-gr-backup*CLI> -- SIP/80620-08296488 is ringing pbx-gr-backup*CLI> <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-iyte6i6dc4j0;received=132.64.4.137;rport=2051 From: "Test" ;tag=07wezivh5v To: ;tag=as1389e95c Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 200 OK Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 102 INVITE From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6e6dd32f;rport Content-Length: 199 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 2129073867 IN IP4 132.64.4.123 s=SIP Call c=IN IP4 132.64.4.123 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (18 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.123:34008 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.123:34008 --- set_address_from_contact host '132.64.4.123' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.123, port 5060 Transmitting (no NAT) to 132.64.4.123:5060: ACK sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK1101ffb9;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Contact: Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> -- SIP/80620-08296488 answered SIP/80609-08280f60 Audio is at 132.64.9.162 port 12624 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-iyte6i6dc4j0;received=132.64.4.137;rport=2051 From: "Test" ;tag=07wezivh5v To: ;tag=as1389e95c Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 706342574 706342574 IN IP4 132.64.4.123 s=session c=IN IP4 132.64.4.123 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> ACK sip:80620@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-0q7ttpr6oxou;rport From: "Test" ;tag=07wezivh5v To: ;tag=as1389e95c Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-u0jdk9n10u9x;rport From: ;tag=2iswzd9g6i To: Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 3 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2051 <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-u0jdk9n10u9x;received=132.64.4.137;rport=2051 From: ;tag=2iswzd9g6i To: ;tag=as32b1bb3d Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 3 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="1d0fae7a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-elsewifmp8zf;rport From: ;tag=2iswzd9g6i To: Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 4 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80609",realm="cc.huji.ac.il",nonce="1d0fae7a",uri="sip:80620@pbx-gr-backup.cc.huji.ac.il;user=phone",response="b847e521e281a9c81ac15994ad8bb8bf",algorithm=md5 Expires: 0 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80609' for '80609' from 132.64.4.137:2051 Looking for 80620 in huji-remote-gr (domain pbx-gr-backup.cc.huji.ac.il) [Sep 6 08:12:52] NOTICE[29379]: chan_sip.c:15967 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-elsewifmp8zf;received=132.64.4.137;rport=2051 From: ;tag=2iswzd9g6i To: ;tag=as32b1bb3d Call-ID: 3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4 CSeq: 4 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c2f74fd5302-6o3i5qco587u@snom320-0004132480B4' Method: SUBSCRIBE pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> REFER sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-vrqaw2oslnya;rport From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 2 REFER Max-Forwards: 70 Contact: ;flow-id=1 Refer-To: sip:80620@132.64.9.162?Replaces=3c2f74fc5091-7j8gfdau0kzd%40snom320-0004132480B4%3Bto-tag%3Das1389e95c%3Bfrom-tag%3D07wezivh5v Referred-By: sip:80609@pbx-gr-backup.cc.huji.ac.il User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 80620@huji-remote-gr by 80609@pbx-gr-backup.cc.huji.ac.il <--- Transmitting (NAT) to 132.64.4.137:2051 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-vrqaw2oslnya;received=132.64.4.137;rport=2051 From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 2 REFER User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/gr-pbx-link-08222fe8 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Reliably Transmitting (NAT) to 132.64.4.137:2051: NOTIFY sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6abb62dc;rport Max-Forwards: 70 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Contact: Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r81632M Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- INVITESTATE:5 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.123, port 5060 pbx-gr-backup*CLI> Audio is at 132.64.9.162 port 1920 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.123:5060: INVITE sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK28d63e3a;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Contact: Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "7106" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 250 v=0 o=root 1336648888 1336648889 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 53624 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - pbx-gr-backup*CLI> a=ptime:20 a=sendrecv --- INVITESTATE:6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Audio is at 132.64.9.162 port 41562 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:7106@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK7093ad22;rport Max-Forwards: 70 From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Contact: Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "Yehavi Bourvine" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 248 v=0 o=root 644232439 644232440 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 49670 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr-backup*CLI> Scheduling destruction of SIP dialog '3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Reliably Transmitting (no NAT) to 132.64.4.137:2051: BYE sip:80609@132.64.4.137:2051;line=k5jv6xlx SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6a91be5a;rport Max-Forwards: 70 From: ;tag=as1389e95c To: "Test" ;tag=07wezivh5v Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> Scheduling destruction of SIP dialog '6242f7fc7e57764f21bfa8841b57da47@132.64.9.162' in 32000 ms (Method: REFER) == Spawn extension (huji-local, 80609, 23) exited non-zero on 'SIP/80609-08280f60' pbx-gr-backup*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK7093ad22;rport From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Date: Thu, 06 Sep 2007 05:29:25 GMT Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK7093ad22;rport From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Date: Thu, 06 Sep 2007 05:29:25 GMT Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 4248 6846 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 18260 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (15 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 128.139.26.6:18260 Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 128.139.26.6:18260 --- set_address_from_contact host '128.139.26.6' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:7106@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK51ec4b33;rport Max-Forwards: 70 From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Contact: Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6abb62dc;rport=5060 From: "7106" ;tag=as34b7f090 To: "Test" ;tag=bd1wdy40ux Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 103 NOTIFY Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 100 Trying Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 103 INVITE From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK28d63e3a;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (8 headers 0 lines) --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> SIP/2.0 200 OK Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 103 INVITE From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK28d63e3a;rport Content-Length: 199 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 2129073867 IN IP4 132.64.4.123 s=SIP Call c=IN IP4 132.64.4.123 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (18 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.123:34008 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.123:34008 --- set_address_from_contact host '132.64.4.123' set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.123, port 5060 Transmitting (no NAT) to 132.64.4.123:5060: ACK sip:80620@132.64.4.123:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK2a355b69;rport Max-Forwards: 70 From: "Yehavi SNOM 320" ;tag=as56538c2b To: ;tag=b53bbc900889c36 Contact: Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK6a91be5a;rport=5060 From: ;tag=as1389e95c To: "Test" ;tag=07wezivh5v Call-ID: 3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4 CSeq: 102 BYE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3c2f74fc5091-7j8gfdau0kzd@snom320-0004132480B4' Method: ACK pbx-gr-backup*CLI> <--- SIP read from 132.64.4.137:2051 ---> BYE sip:7106@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-gh90mwlzehge;rport From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 132.64.4.137 : 2051 (NAT) Scheduling destruction of SIP dialog '6242f7fc7e57764f21bfa8841b57da47@132.64.9.162' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-gh90mwlzehge;received=132.64.4.137;rport=2051 From: "Test" ;tag=bd1wdy40ux To: "7106" ;tag=as34b7f090 Call-ID: 6242f7fc7e57764f21bfa8841b57da47@132.64.9.162 CSeq: 3 BYE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> <--- SIP read from 132.64.4.123:5060 ---> BYE sip:80609@132.64.9.162 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.123:5060;branch=z9hG4bK51a7e6ecb Max-Forwards: 70 Content-Length: 0 To: "Yehavi SNOM 320" ;tag=as56538c2b From: ;tag=b53bbc900889c36 Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 256908126 BYE Supported: timer Supported: replaces User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (11 headers 0 lines) --- Sending to 132.64.4.123 : 5060 (no NAT) <--- Transmitting (no NAT) to 132.64.4.123:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.123:5060;branch=z9hG4bK51a7e6ecb;received=132.64.4.123 From: ;tag=b53bbc900889c36 To: "Yehavi SNOM 320" ;tag=as56538c2b Call-ID: 0cd68f5f35b00109702e54b47e0e5278@132.64.9.162 CSeq: 256908126 BYE User-Agent: Asterisk PBX SVN-trunk-r81632M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr-backup*CLI> == Spawn extension (huji-remote-gr, 80620, 23) exited non-zero on 'SIP/gr-pbx-link-08222fe8' Scheduling destruction of SIP dialog 'F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Reliably Transmitting (no NAT) to 128.139.26.6:5060: BYE sip:7106@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK1a55b480;rport Max-Forwards: 70 From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r81632M Content-Length: 0 --- pbx-gr-backup*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.162:5060;branch=z9hG4bK1a55b480;rport From: ;tag=as04f0a552 To: ;tag=24257ABC-654 Date: Thu, 06 Sep 2007 05:29:27 GMT Call-ID: F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '0cd68f5f35b00109702e54b47e0e5278@132.64.9.162' Method: BYE Really destroying SIP dialog 'F2574180-5B7011DC-993AE2AA-1C68A504@128.139.26.6' Method: ACK pbx-gr-backup*CLI> quit