Re: WebSocket Stasis Control Best Practice

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



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@default:1] Answer("SIP/flowroute-00000012", "") in new stack
    -- Executing [amdme@default:2] Set("SIP/flowroute-00000012", "TALK_DETECT(set)=1000") in new stack
    -- Executing [amdme@default:3] Monitor("SIP/flowroute-00000012", "wav,recordme") in new stack
    -- Executing [amdme@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@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)

-- 
KB 
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

_______________________________________________
asterisk-app-dev mailing list

_______________________________________________
asterisk-app-dev mailing list
asterisk-app-dev@xxxxxxxxxxxxxxxx
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev

[Index of Archives]     [Asterisk SS7]     [Asterisk Announcements]     [Asterisk Users]     [PJ SIP]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [ECOS]     [Linux API]

  Powered by Linux