[PJSIP] Requirement of "line=yes" in ITSP peering/incoming?

What does “line” mean in terms of registration to an ITSP here the definition is given as

Isn’t that exactly what I want? But none of the examples give this. For example, nothing in res_pjsip+Configuration+Examples nor in Endpoints and Location, A Match Made in Heaven nor in Asterisk 13 Configuration_res_pjsip so I’m a bit confused!

But it is mentioned in wiki.asterisk.org/wiki/display/ … gistration

[quote]; If you would like to enable line support and have incoming calls related to this
; registration go to an endpoint automatically the “line” and “endpoint” options must
; be set. The “endpoint” option specifies what endpoint the incoming call should be
; associated with.[/quote]

I was wondering if this was why incoming calls were getting cancelled, but even with and without these config settings, the call never gets to the dialplan.

A quick search suggests that it may be a proprietary extension to SIP, which, if implemented by the peer, allows you to have multiple registrations to the same peer and with either the same contact user, or with the peer overriding the contact user information (e.g. with incoming dialed digits), and still being able to distinguish between calls associated with different registrations. If so, it will be dependent on the peer supporting the feature.

The Contact header provided when registering includes a parameter when “line=yes” is set. This parameter is supposed to be provided when we are called. The remote side may or may not do this. Without the parameter there is no identifying information to associate the INVITE back to the registration.