I’m trying to move a SIP trunk from chan_sip to chan_pjsip.
Everything looks good, registration, incoming calls, etc. But I can’t make any outgoing calls.
The provider support claims the issue is the Contact header, when I’m registering with chan_pjsip there is additional note at the end like:
Wow, you did it!
It get registered without this ‘line’, but I guess I have to find a way to do it through FreePBX because now Asterisk answer Forbidden toany OPTIONS from the sip trink provider and when trying to make outgoing call I get
Unable to create PJSIP channel - endpoint 'XXX' was not found.