So i hope this is just me not configuring things correctly, but i am finding it impossible to get outbound SIP INVITE’s to contain the correct information.
I am attempting to set up a SIP trunk to my ITSP but the calls are failing with an explicit Q850 “User not Found” message. I am using asterisk 11.6.
Here is my SIP config, for obvious reasons i have masked IP address and authentication fields:
[toProvider] type=peer host=10.XXX.XXX.XXX context=default disallow=all allow=ulaw canreinvite=no defaultuser=myusername remotesecret=mysecret fromdomain=mydomain.net
I have checked and double checked these credentials, however i dont think the problem is mismatched credentials at all. Look at the INVITES asterisk sends when using this trunk:
INVITE sip:00XXXXXXXXXX@10.XXX.XXX.XXX SIP/2.0 Via: SIP/2.0/UDP 10.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK31e8a5d2 Max-Forwards: 70 From: "Anonymous" <sip:email@example.com>;tag=as2bf1b624 To: <sip:00XXXXXXXXXX@10.XXX.XXX.XXX> Contact: <sip:anonymous@10.ZZZ.ZZZ.ZZZ:5060> Call-ID: firstname.lastname@example.org CSeq: 102 INVITE User-Agent: Asterisk PBX 11.6.0 Date: Sun, 23 Feb 2014 15:59:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Again i have masked certain details for obvious reason, but this is an actual INVITE taken from a SIP debug. In the above section 00XXXXXXXXXX is the number i am calling, 10.XXX.XXX.XXX is the IP address of my providers SIP gateway (as configured in my peer definition, see above) and 10.ZZZ.ZZZ.ZZZ is my asterisk server.
You can clearly see that the From: header doesnt contain ANY of the authentication credentials i have set in the SIP config. I took these settings from the example sip.conf provided with asterisk, so i hope they are the correct ones.
Have i missed something out?