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Summary:ASTERISK-25507: Dual channel redirect causes Volume audiohook to fail to work
Reporter:James Terhune (bigjimmy)Labels:
Date Opened:2015-10-29 13:54:42Date Closed:2017-03-21 08:49:59
Priority:MinorRegression?
Status:Closed/CompleteComponents:Applications/app_dial
Versions:11.19.0 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Ubuntu 12.04 / 64bitAttachments:
Description:Summary:

SIP device_A calls a number that has a dialplan that ends with Dial(SIP/device_B)
SIP device_B answers phone
User B uses AMI to set the TX volume of device_B's channel to -9
User B notes decreased volume
User B redirects both channel A and B to dialplan that plays MusicOnHold()
User B notes volume increased to normal
Any subsequent attempts to set the volume on channel B fail to set it.

If the volume is set on channel A, it works as expected (the volume remains low).

It appears the cause is that ast_autoservice_chan_hangup_peer(chan, peer) is being called from app_dial during the redirect. This is causing the status of the audiohook to be set to AST_AUDIOHOOK_STATUS_DONE.
Comments:By: Asterisk Team (asteriskteam) 2015-10-29 13:54:43.542-0500

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: James Terhune (bigjimmy) 2017-03-21 08:49:20.117-0500

I retested this with Asterisk 13.14.0 and it appears to have been resolved. This can be closed.


By: James Terhune (bigjimmy) 2017-03-21 08:49:59.228-0500

Retested on Asterisk 13.14.0, fixed.