rendered paste body<-- SIP read from 212.150.36.116:5060:
INVITE sip:422008213@72.61.82.119 SIP/2.0
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK4d164cc1
From: "00442072009205" <sip:00442072009205@212.150.36.116>;tag=as21074299
To: <sip:422008213@72.61.82.119>
Contact: <sip:00442072009205@212.150.36.116>
Call-ID: 644008c24613d2d24efb401a55be9d57@212.150.36.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 08 Aug 2006 20:24:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 372
v=0
o=root 4420 4420 IN IP4 212.150.36.116
s=session
c=IN IP4 212.150.36.116
t=0 0
m=audio 10728 RTP/AVP 18 4 111 3 97 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/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 - - - -
--- (12 headers 16 lines)---
Using INVITE request as basis request - 644008c24613d2d24efb401a55be9d57@212.150.36.116
Sending to 212.150.36.116 : 5060 (non-NAT)
Found peer 'didww-in'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 212.150.36.116:10728
Found description format G729
Found description format G723
Found description format G726-32
Found description format GSM
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 422008213 in from-didww (domain 72.61.82.119)
Reliably Transmitting (no NAT) to 212.150.36.116:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK4d164cc1;received=212.150.36.116
From: "00442072009205" <sip:00442072009205@212.150.36.116>;tag=as21074299
To: <sip:422008213@72.61.82.119>;tag=as3ca0b68a
Call-ID: 644008c24613d2d24efb401a55be9d57@212.150.36.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:422008213@72.61.82.119>
Content-Length: 0
---
Canucks*CLI>
<-- SIP read from 212.150.36.116:5060:
ACK sip:422008213@72.61.82.119 SIP/2.0
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK4d164cc1
From: "00442072009205" <sip:00442072009205@212.150.36.116>;tag=as21074299
To: <sip:422008213@72.61.82.119>;tag=as3ca0b68a
Contact: <sip:00442072009205@212.150.36.116>
Call-ID: 644008c24613d2d24efb401a55be9d57@212.150.36.116
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '644008c24613d2d24efb401a55be9d57@212.150.36.116'
Canucks*CLI>
<-- SIP read from 212.150.36.116:5060:
INVITE sip:422008213@72.61.82.119 SIP/2.0
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK5205153e
From: "18602679312" <sip:18602679312@212.150.36.116>;tag=as144b10e6
To: <sip:422008213@72.61.82.119>
Contact: <sip:18602679312@212.150.36.116>
Call-ID: 19240c79119d19a26199237d5f37df4b@212.150.36.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 08 Aug 2006 20:24:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
ontent-Length: 372
v=0
o=root 4420 4420 IN IP4 212.150.36.116
s=session
c=IN IP4 212.150.36.116
t=0 0
m=audio 13668 RTP/AVP 18 4 111 3 97 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/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 - - - -
--- (12 headers 16 lines)---
Using INVITE request as basis request - 19240c79119d19a26199237d5f37df4b@212.150.36.116
Sending to 212.150.36.116 : 5060 (non-NAT)
Found peer 'didww-in'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 212.150.36.116:13668
Found description format G729
Found description format G723
Found description format G726-32
Found description format GSM
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 422008213 in from-didww (domain 72.61.82.119)
Reliably Transmitting (no NAT) to 212.150.36.116:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK5205153e;received=212.150.36.116
From: "18602679312" <sip:18602679312@212.150.36.116>;tag=as144b10e6
To: <sip:422008213@72.61.82.119>;tag=as573dbec6
Call-ID: 19240c79119d19a26199237d5f37df4b@212.150.36.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:422008213@72.61.82.119>
Content-Length: 0
---
Canucks*CLI>
<-- SIP read from 212.150.36.116:5060:
ACK sip:422008213@72.61.82.119 SIP/2.0
Via: SIP/2.0/UDP 212.150.36.116:5060;branch=z9hG4bK5205153e
From: "18602679312" <sip:18602679312@212.150.36.116>;tag=as144b10e6
To: <sip:422008213@72.61.82.119>;tag=as573dbec6
Contact: <sip:18602679312@212.150.36.116>
Call-ID: 19240c79119d19a26199237d5f37df4b@212.150.36.116
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '19240c79119d19a26199237d5f37df4b@212.150.36.116'
Canucks*CLI>