[asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

[Matthew Jordan]

On Mon, Dec 22, 2014 at 12:53 PM, Joshua Colp <jcolp at digium.com> wrote:
> Currently within PJSIP there is no mechanism for the dialplan to examine
> things at an AOR or contact level. The depth to which you can venture is
> simply the endpoint.
>
> In practice this can be annoying if you would like to examine things and
> change an outgoing INVITE (to add a header for auto answer, for example). To
> aid with this I'd like to propose two new dialplan functions: PJSIP_AOR and
> PJSIP_CONTACT.
>
> The PJSIP_AOR dialplan function would take the name of an AOR and return the
> same information as "pjsip show aor". It would also make available a comma
> separated list of contacts currently bound to the AOR (be it externally
> added or configured).
>
> Assumption:
> An endpoint and AOR exist with the name "blink". A softphone is registered
> to the AOR.
>
> Example:
> ${PJSIP_AOR(blink,mailboxes)} would return "1000" assuming it was configured
> with "1000".
>
> Example:
> ${PJSIP_AOR(blink,contact)} would return
> "blink/sip:38725691 at 192.168.137.1:53015;transport=TCP"
>
> The PJSIP_CONTACT dialplan function would take in the name of a contact as
> returned by PJSIP_AOR. It would make available the same information as
> "pjsip show contact" as well user agent if available.
>
> Example:
> ${PJSIP_CONTACT(blink/sip:38725691 at 192.168.137.1:53015;transport=TCP,user_agent)}
> would return "Blink 0.9.1.2 (Windows)"
>
> While both of these help to query for information if you follow the
> endpoint->aor->contact path there is currently no way, within a pre-dial
> handler, to know what contact (if any) is being used. I'd like to extend the
> ${CHANNEL()} implementation with two additional options, aor and contact,
> which would provide this information.
>
> Assumption:
> An outgoing channel using Dial(PJSIP/blink) is created.
>
> Example:
> ${CHANNEL(aor)} would return "blink"
>
> Example:
> ${CHANNEL(contact)} would return
> "blink/sip:38725691 at 192.168.137.1:53015;transport=TCP"
>
> This assumes that the channel has been dialed using an AOR which resolved
> into a contact. If an explicit contact has been provided (which is done with
> PJSIP_DIAL_CONTACTS) then only contact would return a value.
>
> This would allow the pre-dial handler to query for additional information
> about the contact being dialed in order to do things.
>

I think this gets 90% of the way towards what people (such as X-Rob,
who filed ASTERISK-24341) were looking for.

I don't think this quite addresses the use case of "I want to create
an outbound INVITE request to all contacts on an AoR that do not
currently have an active channel/media session associated with them" -
since we have no way to query through the CHANNEL function what
previous dial attempts resulted in.

It feels like there should be a mechanism that we could provide that
would have this information:
 * We know what contacts we are going to send outbound INVITE requests to
 * We know when those INVITE requests succeed and/or fail
 * And we know when the overall dialog established by those INVITE
requests are done

What technical hurdles do you see with providing this?

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org