sip trunk config and sip debug output

trunk config in the sip.conf file

[trunk-smsmedia-warid]
type=friend
host=10.1.16.72
context=to-warid
disallow=all
allow=gsm                       ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
nat = no
canreinvite=no
port=5060

outbound calls context in the extension.conf file

[to-warid]
exten => 0800707020,1,Dial(SIP/trunk-smsmedia-warid/${EXTEN}, 60)
exten => 0800707020,n,Hangup()

sip debug output

baraza*CLI> console dial 0800707020@to-warid
Audio is at 172.18.0.34 port 13374
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.16.72:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as31a21a46
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Thu, 05 May 2011 16:08:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 471337243 471337243 IN IP4 172.18.0.34
s=Asterisk PBX 1.6.2.18
c=IN IP4 172.18.0.34
t=0 0
m=audio 13374 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[May  5 19:08:14] NOTICE[2986]: console_video.c:133 console_video_start: voice only, console video support not present

<--- SIP read from UDP:10.1.16.72:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport=5060
Call-ID: [email protected]
From: "asterisk"<sip:[email protected]>;tag=as31a21a46
To: <sip:[email protected]:5060>
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:10.1.16.72:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport=5060
Call-ID: [email protected]
From: "asterisk"<sip:[email protected]>;tag=as31a21a46
To: <sip:[email protected]:5060>;tag=6c0828e6
CSeq: 102 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.1.16.72:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as31a21a46
To: <sip:[email protected]:5060>;tag=6c0828e6
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.18
Content-Length: 0


---
 << Hangup on console >>
Really destroying SIP dialog '[email protected]' Method: INVITE