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List:       asterisk-dev
Subject:    Re: [asterisk-dev] pjsip vs cel
From:       Matthew Jordan <mjordan () digium ! com>
Date:       2015-06-11 13:53:56
Message-ID: CAN2PU+62x2bfHW1Sfgc8q9ust4qEO2mM7Fp1nBw9GzN=rvnvyw () mail ! gmail ! com
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On Wed, Jun 10, 2015 at 11:30 PM, James Cloos <cloos@jhcloos.com> wrote:
>>>>>> "RM" == Richard Mudgett <rmudgett@digium.com> writes:
>
> RM> You can specify the dialplan context incoming calls go to when defining
> RM> the endpoint.
>
> Obviously I've done that.  The issue is that the CHAN_START cel event
> does not reflect the specified context and the INVITE's ruri like
> chan_sip's CHAN_START event does, even though the subsequent events do.
>
> I thought I had explained that clearly; apologies for missing any
> ambiguity in my note.

It is a bug. When we allocate a channel in chan_pjsip, we are passing
empty strings into ast_channel_alloc_with_endpoint for the extension
and context parameters. The act of creating the channel in
ast_channel_alloc will, when the routine is finished, publish the
existence of the channel to the 'world' via Stasis, which will create
the CHAN_START event in CEL.

We set the context/extension later on in chan_pjsip_new after the
channel has been created; it should be trivial to refactor that to
pass that information into ast_channel_alloc_with_endpoint.


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

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