[root@asterisk1 asterisk]# asterisk -rvvvvvvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.26.1 svn rev 79171, Copyright (C) 1999 - 2007 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 1.2.26.1 svn rev 79171 currently running on asterisk1 (pid = 3026) Verbosity was 3 and is now 9 asterisk1*CLI> sip debug SIP Debugging re-enabled 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.1.105:2060: OPTIONS sip:105@192.168.1.105:2060;line=d8vbfxfg SIP/2.0 Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK14cceb61;rport From: "Unknown" ;tag=as180c9dfa To: Contact: Call-ID: 2d6a2ded0dd4d74e34d215b31d501472@192.168.1.9 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 19 Feb 2008 19:25:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.1.105:2060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK14cceb61;rport=5060 From: "Unknown" ;tag=as180c9dfa To: Call-ID: 2d6a2ded0dd4d74e34d215b31d501472@192.168.1.9 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom300/6.0.3 Accept-Language: en 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 Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '2d6a2ded0dd4d74e34d215b31d501472@192.168.1.9' asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: OPTIONS sip:metaswitch@72.18.39.109:5060;transport=udp SIP/2.0 To: From: ;tag=metaswitch.newroadstelecom.net+1+0+3509dc27 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-bab6db57fd03bc7a0a35-1--d87543- Call-ID: AAC288AC@metaswitch.newroadstelecom.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom300/6.0.3 Accept-Language: en 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 Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '2d6a2ded0dd4d74e34d215b31d501472@192.168.1.9' asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: OPTIONS sip:metaswitch@72.18.39.109:5060;transport=udp SIP/2.0 To: From: ;tag=metaswitch.newroadstelecom.net+1+0+3509dc27 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-bab6db57fd03bc7a0a35-1--d87543- Call-ID: AAC288AC@metaswitch.newroadstelecom.net CSeq: 592510977 OPTIONS Contact: Max-Forwards: 69 Organization: Supported: 100rel, 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 0 --- (12 headers 0 lines) --- Looking for metaswitch in from-sip-external (domain 72.18.39.109) Transmitting (NAT) to 209.183.160.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-bab6db57fd03bc7a0a35-1--d87543-;received=209.183.160.130 From: ;tag=metaswitch.newroadstelecom.net+1+0+3509dc27 To: ;tag=as0ee1aaaa Call-ID: AAC288AC@metaswitch.newroadstelecom.net CSeq: 592510977 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'AAC288AC@metaswitch.newroadstelecom.net' Destroying call '3f7d060538ef2cb9087723dc3172300c@127.0.0.1' asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: INVITE sip:4792422484@72.18.39.109:5060;transport=udp SIP/2.0 To: From: Anonymous;tag=c12be624 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-21880c43d2d50c1ee896-1-cHAxNzhlYzMwZTcwODQxMDQwZmM5Zg..-d87543- Call-ID: 9d3535cf666a51f27a752c98b716e1cf CSeq: 77380486 INVITE Contact: Anonymous Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 169 v=0 o=- 858285753 829944631 IN IP4 209.183.160.130 s=- c=IN IP4 209.183.160.130 t=0 0 m=audio 58112 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines) --- Using INVITE request as basis request - 9d3535cf666a51f27a752c98b716e1cf Sending to 209.183.160.130 : 5060 (NAT) Found peer 'NewRoads-Trunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.160.130:58112 Found description format telephone-event Capabilities: us - 0x0 (nothing), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Transmitting (NAT) to 209.183.160.130:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-21880c43d2d50c1ee896-1-cHAxNzhlYzMwZTcwODQxMDQwZmM5Zg..-d87543-;received=209.183.160.130 From: Anonymous;tag=c12be624 To: ;tag=as3c027c3d Call-ID: 9d3535cf666a51f27a752c98b716e1cf CSeq: 77380486 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: ACK sip:4792422484@72.18.39.109:5060;transport=udp SIP/2.0 To: ;tag=as3c027c3d From: Anonymous;tag=c12be624 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-21880c43d2d50c1ee896-1-cHAxNzhlYzMwZTcwODQxMDQwZmM5Zg..-d87543- Call-ID: 9d3535cf666a51f27a752c98b716e1cf CSeq: 77380486 ACK Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '9d3535cf666a51f27a752c98b716e1cf' asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: INVITE sip:4792422484@72.18.39.109:5060;transport=udp SIP/2.0 To: From: Anonymous;tag=cc14c823 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-cc03ab74affde33bd9b0-1-cHA3ZjY0ZjgwYzkwN2FmZjBhNGJkNQ..-d87543- Call-ID: 8d1c93419ef786a8c039cee82dc60892 CSeq: 71245959 INVITE Contact: Anonymous Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 169 v=0 o=- 858287353 829946539 IN IP4 209.183.160.130 s=- c=IN IP4 209.183.160.130 t=0 0 m=audio 61304 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines) --- Using INVITE request as basis request - 8d1c93419ef786a8c039cee82dc60892 Sending to 209.183.160.130 : 5060 (NAT) Found peer 'NewRoads-Trunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.160.130:61304 Found description format telephone-event Capabilities: us - 0x0 (nothing), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Transmitting (NAT) to 209.183.160.130:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-cc03ab74affde33bd9b0-1-cHA3ZjY0ZjgwYzkwN2FmZjBhNGJkNQ..-d87543-;received=209.183.160.130 From: Anonymous;tag=cc14c823 To: ;tag=as1dd61a8b Call-ID: 8d1c93419ef786a8c039cee82dc60892 CSeq: 71245959 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 209.183.160.130:5060: ACK sip:4792422484@72.18.39.109:5060;transport=udp SIP/2.0 To: ;tag=as1dd61a8b From: Anonymous;tag=cc14c823 Via: SIP/2.0/UDP 209.183.160.130:5060;branch=z9hG4bK-d87543-cc03ab74affde33bd9b0-1-cHA3ZjY0ZjgwYzkwN2FmZjBhNGJkNQ..-d87543- Call-ID: 8d1c93419ef786a8c039cee82dc60892 CSeq: 71245959 ACK Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '8d1c93419ef786a8c039cee82dc60892' asterisk1*CLI>