rendered paste body<-- SIP read from 4.79.19.58:5060:
ACK sip:3145581766@216.178.27.13 SIP/2.0
Via: SIP/2.0/UDP 4.79.19.58:5060;branch=z9hG4bK7ad82667;rport
From: "3147811800" <sip:3147811800@4.79.19.58>;tag=as35bd8a56
To: <sip:3145581766@216.178.27.13>;tag=as741fc7a4
Contact: <sip:3147811800@4.79.19.58>
Call-ID: 731431d80bae3a5071c5e08b246cd17b@4.79.19.58
CSeq: 102 ACK
User-Agent: Telasip
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '731431d80bae3a5071c5e08b246cd17b@4.79.19.58'
gw01*CLI>
<-- SIP read from 4.79.19.58:5060:
INVITE sip:3145581766@216.178.27.13 SIP/2.0
Via: SIP/2.0/UDP 4.79.19.58:5060;branch=z9hG4bK5084e206;rport
From: "3147811800" <sip:3147811800@4.79.19.58>;tag=as36282ddc
To: <sip:3145581766@216.178.27.13>
Contact: <sip:3147811800@4.79.19.58>
Call-ID: 540c752e6de5dfaa6e037dfc291a9626@4.79.19.58
CSeq: 102 INVITE
User-Agent: Telasip
Date: Mon, 12 Jun 2006 21:28:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 342
v=0
o=root 31241 31241 IN IP4 4.79.19.58
s=session
c=IN IP4 4.79.19.58
t=0 0
m=audio 30492 RTP/AVP 0 18 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
m=video 12906 RTP/AVP 34
a=rtpmap:34 H263/90000
--- (12 headers 15 lines)---
Using INVITE request as basis request - 540c752e6de5dfaa6e037dfc291a9626@4.79.19.58
Sending to 4.79.19.58 : 5060 (NAT)
Found peer 'telasip-out'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 111
Found RTP audio format 101
Peer audio RTP is at port 4.79.19.58:30492
Found description format PCMU
Found description format G729
Found description format GSM
Found description format G726-32
Found description format telephone-event
Found description format H263
Capabilities: us - 0x50c (ulaw|alaw|g729|ilbc), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 3145581766 in eci-incoming (domain 216.178.27.13)
Reliably Transmitting (no NAT) to 4.79.19.58:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 4.79.19.58:5060;branch=z9hG4bK5084e206;rport;received=4.79.19.58
From: "3147811800" <sip:3147811800@4.79.19.58>;tag=as36282ddc
To: <sip:3145581766@216.178.27.13>;tag=as2f87eaaf
Call-ID: 540c752e6de5dfaa6e037dfc291a9626@4.79.19.58
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3145581766@216.178.27.13>
Content-Length: 0
---
gw01*CLI>
<-- SIP read from 4.79.19.58:5060:
ACK sip:3145581766@216.178.27.13 SIP/2.0
Via: SIP/2.0/UDP 4.79.19.58:5060;branch=z9hG4bK5084e206;rport
From: "3147811800" <sip:3147811800@4.79.19.58>;tag=as36282ddc
To: <sip:3145581766@216.178.27.13>;tag=as2f87eaaf
Contact: <sip:3147811800@4.79.19.58>
Call-ID: 540c752e6de5dfaa6e037dfc291a9626@4.79.19.58
CSeq: 102 ACK
User-Agent: Telasip
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '540c752e6de5dfaa6e037dfc291a9626@4.79.19.58'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 4.79.19.58:5060:
OPTIONS sip:gw3.telasip.com SIP/2.0
Via: SIP/2.0/UDP 216.178.27.13:5060;branch=z9hG4bK067865fe;rport
From: "Unknown" <sip:Unknown@216.178.27.13>;tag=as15f63787
To: <sip:gw3.telasip.com>
Contact: <sip:Unknown@216.178.27.13>
Call-ID: 60e4a68004a9b81d52ef21ad6af5ea62@216.178.27.13
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 12 Jun 2006 21:29:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
gw01*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.178.27.13:5060;branch=z9hG4bK067865fe
From: "Unknown" <sip:Unknown@216.178.27.13>;tag=as15f63787
To: <sip:gw3.telasip.com>;tag=as5da0d66f
Call-ID: 60e4a68004a9b81d52ef21ad6af5ea62@216.178.27.13
CSeq: 102 OPTIONS
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4.79.19.58>
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '60e4a68004a9b81d52ef21ad6af5ea62@216.178.27.13'
gw01*CLI> sip no debug