[asterisk-app-dev] WebSocket Stasis Control Best Practice
Matthew Jordan
mjordan at digium.com
Tue Jul 8 15:08:26 CDT 2014
On Tue, Jul 8, 2014 at 1:40 PM, Krandon <krandon.bruse at gmail.com> wrote:
> Hey guys,
>
> Scale testing has been going very well. I will have some prelim numbers
> soon. You guys are on top of it. I found one small bug - a crash related to
> string length of the request. I went to Jira to report it, updated Asterisk
> from 3 weeks ago to now, saw there was a patch and the bug was already
> fixed. A+!
>
> I do have a implementation question. I am currently using Websockets to
> create the call request and dump it into Stasis. However, if the initial leg
> A call fails (for whatever reason) then I never see the call come into the
> Stasis app. This is to be expected, as the App has not been invoked. What's
> the best way to get the status of that first call? (SIP code would be great,
> but not necessary first time around)
>
Are you Originating a Local channel from ARI?
If so, we just made a bug fix (going out in 12.4.0) where the half of
the Local channel that goes off into dialplan will also be subscribed
to. The other half will always go into the Stasis application first,
so that gives you something that you're subscribed to that is *also*
doing your dialplan call.
If calling something directly (like a PJSIP endpoint or a SIP peer),
then I'd expect you to be subscribed to that channel if you told it
you wanted it in your Stasis app when it answered. That should give
you a ChannelDestroyed event at the very least - if not, that sounds
buggy.
--
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
More information about the asterisk-app-dev
mailing list