13.11.2/PJSIP: Identify endpoint for inbound calls

[rakanishu]

It appears as if the documentation on that matter on the wiki here https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships is not quite correct. It states:

Section Names:
In most cases, you can name a section whatever makes sense to you. For example you might name a transport [transport-udp-nat] to help you remember how that section is being used.However, in some cases, (endpoint and aor types) the section name has a relationship to its function. In the case of endpoint and aor their names must match the user portion of the SIP URI in the “To” header for inbound SIP requests. The exception to that rule is if you have an identify section configured for that endpoint. In that case the inbound request would be matched by IP instead of against the user in the “To” header.

However I even went as far as to look at the source code, and the only thing it ever seems to match upon is the “From:” header, not the “To:” header.

Given the fact that the endpoint identifiers in pjsip are modular, I still think there should be a way to match on IP address first and “To:” header second to identify the correct endpoint for inbound calls in a scenario where you register multiple DIDs with the same ITSP. A similar issue has been discussed here https://community.asterisk.org/t/sip-header-did-help/66528/2, so maybe it’s not as uncommon as I thought? Personally I feel that neither the use of the ‘line’ parameter, nor reyling on some header parsing magic in the dialplan appears to be the correct approach to solving this kind of problem.