I have configured an extension (10000) with follow-me. In the follow-me list I have written a PSTN number (followed by #).
Then I have created
a SIP trunk, which is registered to a SIP provider (rebvoice) to give access to the PSTN.
an outbound route matching any dial pattern, linked to the SIP trunk
In the asterisk console I could see (with “pjsip show registrations”) that the trunk is successfully registered to rebvoice.
Then I made a call to the extension 10000, and I would expect that the call is forwarded to the PSTN number. In the full log I can see the INVITE to the rebvoice provider; however the forwarding to the PSTN fails.
Which is not surprising, as there is no SDP payload! The only thing I can think of that might cause that is either specifying allow=all [codecs], although I thought that bug had been fixed (although it is still not a good think to do), or disallow=all with no codecs then allowed (although I would have thought it would have send SDP with no streams, rather than not sent it).
Sorry, I don’t find the allow/disallow field anywhere in the trunk settings. Are you referring to the CODEC page of the trunk, containing a list of checkboxes? They are all checked from the first one until the mpeg4 included. The g723 and the following ones are unchecked.
I realized that the problem was in the dial patterns settings of the outbound route.
This was the same problem that was behind this other issue:
In concrete, I solved it setting a dial pattern with a “.”.
The only remaining problem is that it is a bit hard to tune the initial ring time and ring time, in order to offer a smooth experience to the caller, but this is another story. I will take some time to test and tune it.