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Summary:ASTERISK-26709: res_ari: ARI Snoop-whisper fails (no transmit audio ) to channel in holding bridge
Reporter:Lucas SOLER (lucas.soler)Labels:
Date Opened:2017-01-09 08:56:25.000-0600Date Closed:
Priority:MajorRegression?
Status:Open/NewComponents:Bridges/bridge_holding Core/Bridging
Versions:13.0.0 13.9.1 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Attachments:( 0) debug_log_snooping.txt
( 1) pjsip.conf
( 2) sip.conf
Description:Hi,

I am working on an ACD system and I have a problem that I do not understand.

I have two channels speaking in one bridge (a customer and an agent).
A third person (a supervisor) can snoop and whisper on the agent channel.

Everything works fine. But if the agent holds the caller and the supervisor starts to whisper at this time, the agent cannot hear the supervisor.

When agent stops the holding, he hears the voice of the supervisor with one minute of delay (the holding time).

The agent (dialed channel) and the caller (stasis channel) are in a mixing bridge. The agent (snooped channel) and the supervisor (snooper channel) are in a second mixing bridge.
When the agent starts holding the caller's channel, I remove it from the bridge, put it in an holding bridge and finally start music on hold.
The supervior who is snooping is mute because I don't want the agent to know that someone is listening. When I start whispering, I unmute the snooping channel.
When agent stops the holding, I stop the moh, remove the channel from the holding bridge and add it in the bridge with the agent (snooped by the supervisor).

I am using ARI with the client for javascript (node-ari-client) and tested it on versions 13.0 and 13.9 of Asterisk.

You will find in the URL a github repository with a light version of the functionnality written in javascript.

Thank you
Comments:By: Asterisk Team (asteriskteam) 2017-01-09 08:56:26.145-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

By: Rusty Newton (rnewton) 2017-01-09 20:08:44.713-0600

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Lucas SOLER (lucas.soler) 2017-01-10 04:16:55.392-0600

Hi, I uploaded the additional debug.

By: Rusty Newton (rnewton) 2017-01-10 13:29:02.035-0600

Can you provide an pjsip conf for use with the script that reproduces the issue?

Thanks!

By: Lucas SOLER (lucas.soler) 2017-01-11 02:16:54.569-0600

Hello Rusty, we use a sip trunk and chan_sip. I will upload the sip.conf

By: Rusty Newton (rnewton) 2017-01-11 14:30:38.261-0600

Okay, do you know if the issue occurs with pjsip channels as well? I imagine it would, but it is good to be sure.

I ask because chan_sip is no longer core supported, it is under extended support since Asterisk 12.

By: Lucas SOLER (lucas.soler) 2017-01-12 03:07:34.486-0600

I tried it and the issue occurs too. We take notes that we have to migrate to pjsip ;)
You will find the pjsip.conf in attachments.

By: Rusty Newton (rnewton) 2017-01-12 09:16:10.133-0600

Thanks! :)

By: Lucas SOLER (lucas.soler) 2017-03-21 10:18:56.878-0500

Hi Rusty, have you got some news about this issue ? Thank you

By: Joshua C. Colp (jcolp) 2017-03-21 10:24:03.803-0500

This issue was triaged and accepted (as you can see by the Open status). Any updates or further questions will be posted here.

By: Lucas SOLER (lucas.soler) 2017-03-21 10:30:00.198-0500

Ok I am available if you have further questions