I was using asterisk 1.4 and it was working fine. I just upgraded to asterisk 188.8.131.52 and I am getting trouble when receiving some type of calls.
basically, I have the following in sip.conf:
and the following context in extensions.conf:
exten => MyPhoneNumber,1,Dial(SIP/MyPhone)
exten => s,1,Dial(SIP/MyPhone)
When someone dials my phone number from the PSTN network, it goes through my VoIP provider, gets routed just fine to my Asterisk PBX, and my phone rings. This is the expected behavior, since my SIP provider calls sip:MyPhoneNumber@MyIPAddressHere
The problem is, my VoIP provider also give me internal extensions. I assigned the internal extension 1000 to that Asterisk server. When I call 1000 from an other phone, however, the SIP call that I receive is to sip:MyIPAddressHere (that’s right, there is no username in that SIP uri). Weird behavior, which mean that the 1000 extension isn’t used, but there should still be a way to handle this, right?
I tought the “s” extension was supposed to take care of that case, but it doesn’t since I receive the following error message:
NOTICE: chan_sip.c:21614 handle_request_invite: Call from ‘voipms’ to extension ‘MyIPAddressHere’ rejected because extension not found in context ‘voipms’.
Again, it was working fine in Asterisk 1.4