rendered paste body[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 <markster@digium.com>
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" <sip:Unknown@192.168.1.9>;tag=as180c9dfa
To: <sip:105@192.168.1.105:2060;line=d8vbfxfg>
Contact: <sip:Unknown@192.168.1.9>
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" <sip:Unknown@192.168.1.9>;tag=as180c9dfa
To: <sip:105@192.168.1.105:2060;line=d8vbfxfg>
Call-ID: 2d6a2ded0dd4d74e34d215b31d501472@192.168.1.9
CSeq: 102 OPTIONS
Contact: <sip:105@192.168.1.105:2060;line=d8vbfxfg>;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: <sip:metaswitch@72.18.39.109>
From: <sip:metaswitch@192.168.12.20:5060;transport=udp>;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: <sip:105@192.168.1.105:2060;line=d8vbfxfg>;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: <sip:metaswitch@72.18.39.109>
From: <sip:metaswitch@192.168.12.20:5060;transport=udp>;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: <sip:3fy3LPaXmRmubYymRpq8_XwYbWRyz6Ji2CaKLo@209.183.160.130:5060>
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: <sip:metaswitch@192.168.12.20:5060;transport=udp>;tag=metaswitch.newroadstelecom.net+1+0+3509dc27
To: <sip:metaswitch@72.18.39.109>;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: <sip:72.18.39.109>
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: <sip:4792422484@72.18.39.109>
From: Anonymous<sip:anonymous@192.168.12.20:5060;transport=udp>;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<sip:FJk-5T9ehGCCC8MKOlpJnYbLuIWm_hO80nkWcIi_AA@209.183.160.130:5060>
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<sip:anonymous@192.168.12.20:5060;transport=udp>;tag=c12be624
To: <sip:4792422484@72.18.39.109>;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: <sip:4792422484@72.18.39.109>;tag=as3c027c3d
From: Anonymous<sip:anonymous@192.168.12.20:5060;transport=udp>;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: <sip:4792422484@72.18.39.109>
From: Anonymous<sip:anonymous@192.168.12.20:5060;transport=udp>;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<sip:FJk-5T9ehGCCC8MKOlpJnYbLuIWm_hO80nkWcIi_AA@209.183.160.130:5060>
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<sip:anonymous@192.168.12.20:5060;transport=udp>;tag=cc14c823
To: <sip:4792422484@72.18.39.109>;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: <sip:4792422484@72.18.39.109>;tag=as1dd61a8b
From: Anonymous<sip:anonymous@192.168.12.20:5060;transport=udp>;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>