some of you might have seen my post on the FreePBX-Forum as well, but I didn’t receive any feedback so far. So hopefully here are some guys able to point me to the solution.
My situation: On an All-IP accessI try to connect the Asterisk server to the provider network instead of the heavily scissored CPE.
Incoming calls went fine soon, but outgoing calls is a mess. When trying to place a call, the answer from the provider proxy in the SIP debug is always 403 Forbidden. As I wasn’t able to extract a more detailed reason from the SIP debug, I’ve started a tcpdump and inspected the packets more detailed. I’ve found two major differences in the packets where I wasn’t able to find a solution with Google.
I have modified: UserAgent, SDP session owner, SDP session name
But the “working” SIP trace contains some more information in the SIP-Header and I haven’t found the answer how to modify Asterisk this way:
Route URI: sip:172.x.y.z;lr
Route Host Part: 172.x.y.z
Route URI parameter: lr
And a Contact URI paramter:
Contact URI: sip:email@example.com;uniq=42127CCE7A7A466621DAE696E2365
Contact URI User Part: 2xxx3xxxxxx
Contact URI Host Part: 10.10.x.x
Contact URI parameter: uniq=42127CCE7A7A466621DAE696E2365
Any help is welcom! If you need the whole SIP-capture, please tell me as well