I should probably know this, but I don't.
I have a SIP Trunk provider that we are considering using and we want to test trunks with them. They sent over a form that has lots or questions that I was able to answer pretty easily... except for one.
Please provide an example of your SIP INVITE. (this is all it says, no other info. they do give you a whole page to paste it into)
Anybody know what that would look like on MTE?
Thanks.
SIP Debug
It took me a few minutes to find the SIP Debug commands for the Asterisk 1.6 CLI...
sip set debug on
sip set debug off
I created a route for "77777" and pointed to a test SIP trunk called S1. I placed a test call and captured the following... Just in case anyone else wants to know how to do it. (note: I have used and 'x' to replace some of the phone number and IP details)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 206.173.x.x port 17158
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.199.x.x:5060:
INVITE sip:77777@64.199.x.x SIP/2.0
Via: SIP/2.0/UDP 206.173.x.x:5060;branch=z9hG4bK502e5e58;rport
Max-Forwards: 70
From: "RF Communications" ;tag=as3a044b07
To:
Contact:
Call-ID: 77c9587d4f3fd28c5c1e042e67b7198e@206.173.x.x
CSeq: 102 INVITE
User-Agent: Thirdlane
Remote-Party-ID: "RF Communications" ;privacy=off;screen=no
Date: Thu, 10 Feb 2011 21:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 807078236 807078236 IN IP4 206.173.x.x
s=Asterisk PBX 1.6.2.14
c=IN IP4 206.173.x.x
t=0 0
m=audio 17158 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 77777@S1
that usually means they are asking for a capture so that they can see an invite message.