rendered paste body<-- SIP read from 190.54.21.68:5060:
INVITE sip:25804759@68.178.167.75 SIP/2.0
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK0585034e;rport
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>
Contact: <sip:28551194@190.54.21.68>
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 INVITE
User-Agent: CallMe LatinIP.com
Max-Forwards: 70
Date: Tue, 29 May 2007 02:27:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 7836 7836 IN IP4 190.54.21.68
s=session
c=IN IP4 190.54.21.68
t=0 0
m=audio 13274 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 11 lines)---
Using INVITE request as basis request - 1f473550302c5af830d6324b17c23bc9@190.54.21.68
Sending to 190.54.21.68 : 5060 (NAT)
Found peer '25804759'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 190.54.21.68:13274
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 25804759 in from-sip-external (domain 68.178.167.75)
list_route: hop: <sip:28551194@190.54.21.68>
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK0585034e;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Length: 0
---
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK0585034e;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>;tag=as3f386ead
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Length: 0
---
We're at 68.178.167.75 port 17718
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK0585034e;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>;tag=as3f386ead
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 26036 26036 IN IP4 68.178.167.75
s=session
c=IN IP4 68.178.167.75
t=0 0
m=audio 17718 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
ip-68-178-167-75*CLI>
<-- SIP read from 190.54.21.68:5060:
ACK sip:25804759@68.178.167.75 SIP/2.0
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK74328161;rport
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>;tag=as3f386ead
Contact: <sip:28551194@190.54.21.68>
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 ACK
User-Agent: CallMe LatinIP.com
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines)---
ip-68-178-167-75*CLI>
<-- SIP read from 201.246.76.185:49168:
--- (0 headers 1 lines)---
We're at 68.178.167.75 port 17718
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK0585034e;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as214431d4
To: <sip:25804759@68.178.167.75>;tag=as3f386ead
Call-ID: 1f473550302c5af830d6324b17c23bc9@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 26036 26037 IN IP4 68.178.167.75
s=session
c=IN IP4 68.178.167.75
t=0 0
m=audio 17718 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
ip-68-178-167-75*CLI>