[asterisk-app-dev] WebSocket Stasis Control Best Practice
Matthew Jordan
mjordan at digium.com
Tue Aug 5 09:38:01 CDT 2014
On Fri, Aug 1, 2014 at 10:16 AM, Krandon <krandon.bruse at gmail.com> wrote:
> On Friday, August 1, 2014 at 10:01 AM, Joshua Colp wrote:
>
> Krandon wrote:
>
> Just to reignite this whole thread. We've had great success with Stasis
> so far. We have implemented a (imho better) AMD using TALK_DETECT and
> the ARI events. When using this, it seems like calling /play multiple
> times will just queue up the audio files. This is probably the best use
> case and most common implementation. However, if half-way through our
> third audio file we realize that we've been speaking to a machine this
> whole time and we want to start for the beginning - we thought we would
> use DELETE /playbacks/{playbackId}. If we do that, however, and then
> subsequently try to play audio on the channel (even though the call is
> still connected and in the Stasis app) we get a Channel not found. Is
> this the intended use or even a good use?
>
>
> Stopping a playback, or multiple playbacks in sequence, like that
> shouldn't result in you being unable to do things on the channel. Do you
> have the output of what you are invoking and the log from Asterisk? It
> may be a bug.
>
> -- Executing [amdme at default:1] Answer("SIP/flowroute-00000012", "")
> in new stack
> -- Executing [amdme at default:2] Set("SIP/flowroute-00000012",
> "TALK_DETECT(set)=1000") in new stack
> -- Executing [amdme at default:3] Monitor("SIP/flowroute-00000012",
> "wav,recordme") in new stack
> -- Executing [amdme at default:4] Stasis("SIP/flowroute-00000012", "vb")
> in new stack
> -- <SIP/flowroute-00000012> Playing '178.ulaw' (language 'en')
> > SIP/flowroute-00000012 is now talking
> > SIP/flowroute-00000012 is now silent
> [Aug 1 10:44:20] NOTICE[7444][C-00000030]: res_stasis_playback.c:245
> playback_final_update: 2565557733.refMondwandway: Playback stopped for
> sound:178
> [Aug 1 10:44:23] NOTICE[7444][C-00000030]: res_stasis_playback.c:245
> playback_final_update: 2565557733.refMondwandway: Playback stopped for
> sound:silence/3
> -- Executing [amdme at default:5] Hangup("SIP/flowroute-00000012", "")
> in new stack
>
> This happens after we do a DELETE on /playbacks/{playbackId} (which I use
> the channel ID - I am going to test with a different, fixed ID)
>
>
>
Maybe I misunderstood what you just said, but are you using the channel ID
in the DELETE /playbacks/{playbackId} operation?
If so, you should be using the ID of the playback operation, not the
channel ID. That playback ID should be either (a) passed back to you by the
/play operation (on either the bridge or channel), or provided by you when
the /play operation is invoked.
--
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-app-dev/attachments/20140805/9ab40fe0/attachment.html>
More information about the asterisk-app-dev
mailing list