[Home]

Summary:ASTERISK-27766: The voice recording problem of the call that comes from queue during transmitting
Reporter:Mesut Altürk (multimesut)Labels:
Date Opened:2018-03-23 03:41:29Date Closed:2018-04-02 05:06:19
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_mixmonitor
Versions:13.2.0 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debug_log_123456.txt
Description:Hi,

I have a problem that i can't solve for a long time. Voice record is not handed after transmitting call (as in below scenario). After transmitting call voice record should continue, as a result channel is open. There is no problem if i transmit with "features.conf/atxfer => *2" code then record continues. I guess a patch for MixMonitor can be made.

I am waiting your supports for this issue. Thank You..


Scenario;
Voip Trunk -> Queue 630 -> Extension 800 -> Attended transfer Extension 801

Result;
Voip Trunk -> Extension 800 = Record OK
Extension 800 -> Extension 801 = Record OK
Voip Trunk -> Extension 801 = No Record
Comments:By: Asterisk Team (asteriskteam) 2018-03-23 03:41:31.560-0500

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: Mesut Altürk (multimesut) 2018-03-23 03:44:38.589-0500

log dosyası
By: Joshua C. Colp (jcolp) 2018-03-26 10:08:05.903-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

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




By: Mesut Altürk (multimesut) 2018-03-27 01:11:25.291-0500

I added the debug file.
By: Joshua C. Colp (jcolp) 2018-03-27 05:02:45.122-0500

We also need configuration and specific details to reproduce this.
By: Mesut Altürk (multimesut) 2018-03-30 03:41:08.533-0500

I can send what you need also. I have attached log file.
By: Mesut Altürk (multimesut) 2018-04-02 03:27:28.269-0500

I solved the problem as follows. thank you.

queue.conf -> setinterfacevar = yes