rendered paste body<-- Sip read from 207.174.202.4:5060:UDP
INVITE sip:s@10.10.1.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK451e3224;rport
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
To: <sip:s@10.10.1.16;transport=UDP>
Contact: <sip:5555555555@207.174.202.4>
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 INVITE
User-Agent: "TELIAX GEAR v2.1.0x"
Max-Forwards: 70
Date: Fri, 29 Sep 2006 08:11:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 315
v=0
o=root 724 724 IN IP4 207.174.202.4
s=session
c=IN IP4 207.174.202.4
t=0 0
m=audio 12308 RTP/AVP 0 18 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 14 lines)---
Using INVITE request as basis request - 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
Sending to 207.174.202.4: 5060; transport UDP (NAT)
Found peer 'teliax'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 207.174.202.4:12308
Found description format PCMU
Found description format G729
Found description format G726-32
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in default (domain 10.10.1.16;transport=UDP)
list_route: hop: <sip:5555555555@207.174.202.4>
Transmitting (NAT) to 207.174.202.4:5060:UDP
with fd 16
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK451e3224;received=207.174.202.4;rport=5060
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
To: <sip:s@10.10.1.16;transport=UDP>
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.1.16;transport=UDP>
Content-Length: 0
---
UDP write: fd 16
We're at 10.10.1.16 port 19944
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 207.174.202.4:5060:UDP
with fd 16
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK451e3224;received=207.174.202.4;rport=5060
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
To: <sip:s@10.10.1.16;transport=UDP>;tag=as2448280b
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.1.16;transport=UDP>
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 2846 2846 IN IP4 10.10.1.16
s=session
c=IN IP4 10.10.1.16
t=0 0
m=audio 19944 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
UDP write: fd 16
[Kasterisk1*CLI>
<-- Sip read from 207.174.202.4:5060:UDP
ACK sip:s@10.10.1.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK32002555;rport
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
To: <sip:s@10.10.1.16;transport=UDP>;tag=as2448280b
Contact: <sip:5555555555@207.174.202.4>
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 ACK
User-Agent: "TELIAX GEAR v2.1.0x"
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines)---
[Kasterisk1*CLI>
set_destination: Parsing <sip:5555555555@207.174.202.4> for address/port/transport to send to
Reliably Transmitting (NAT) to 207.174.202.4:5060:UDP
with fd 16
BYE sip:5555555555@207.174.202.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.16:5060;branch=z9hG4bK56f43e31;rport
From: <sip:s@10.10.1.16;transport=UDP>;tag=as2448280b
To: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
Contact: <sip:s@10.10.1.16;transport=UDP>
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
UDP write: fd 16
[Kasterisk1*CLI>
<-- Sip read from 207.174.202.4:5060:UDP
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.16:5060;branch=z9hG4bK56f43e31;received=206.72.67.79;rport=5060
From: <sip:s@10.10.1.16;transport=UDP>;tag=as2448280b
To: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as7d3d3c01
Call-ID: 41c7cfa32510f8127d0e78c04119c564@207.174.202.4
CSeq: 102 BYE
User-Agent: "TELIAX GEAR v2.1.0x"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5555555555@207.174.202.4>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing