Hi all
Outgoing call are not working, icoming working fine.
I’m useing Freepbx 14.0.1.19, Asteriks version 13.17.2 and my SIP Channel driver is chan_pjsip.
If I make a outgoing call, the asterisk is setting in the contact vield the username “asterisk”.
My sip trunk provider says that this is not valid.
But how can I change it?
I tried to change it by setting the field - pjsip settings, advanced, contact user, the SIP trunk username. But if I put this there, then nothing has changed at the next call.
An other way to find a solution was, that I change my SIP Channel driver to chan_sip, then the pbx will no longer send any register to my trunk provider.
Have anybody any idea, what do I make wrong?
Thanks in advanced
Thomas
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:[email protected]:5083 SIP/2.0
Message Header
Via: SIP/2.0/UDP xx.xx.xx.xx:5083;rport;branch=z9hG4bKPj2dccfcff-68c8-4549-b051-d1b856dcb725
From: sip:[email protected];tag=ae49fdf0-2dd5-4230-826d-feb7ccee7118
To: sip:[email protected]
Contact: sip:[email protected]:5083
Contact URI: sip:[email protected]:5083
Call-ID: 68b2025f-e073-41f7-bb2c-a70069053d5a
CSeq: 29351 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.19(13.17.2)
Content-Type: application/sdp
Content-Length: 309
Message Body