| Summary: | ASTERISK-25801: app_mixmonitor: Does not continue after attended transfer | ||||
| Reporter: | Vadim (cron333) | Labels: | |||
| Date Opened: | 2016-02-18 04:12:45.000-0600 | Date Closed: | 2016-03-13 17:05:56 | ||
| Priority: | Major | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | Applications/app_mixmonitor | ||
| Versions: | 13.7.0 13.7.2 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | CentOS release 6.7 (Final) | Attachments: | ( 0) debug_log_audiohook.txt | ||
| Description: | The problem is the MixMonitor and the AttendedTransfer.
The scenario: 1) A calls to B (with the UA Zoiper) 2) B answers (start MixMonitor (the MixMonitor command with the option 'b')) 3) A presses Hold 4) A calls to C 5) C answers (start MixMonitor (the MixMonitor command with the option 'b')) 6) After that A makes "Attended Transfer" 7) The call is continuing, but the MixMonitor on both channels are ending In the version 11.x.x it works correctly (with AUDIOHOOK_INHERIT) ... | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2016-02-18 04:12:47.290-0600 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: Joshua C. Colp (jcolp) 2016-02-18 05:53:36.024-0600 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: Rusty Newton (rnewton) 2016-02-21 22:49:13.106-0600 This sounds like a possible duplicate of ASTERISK-25674 - that is, MixMonitor is probably working as expected. We definitely need the information Josh requested. We need to see a verbose and debug log as well as the dialplan involved. It is important to understand *exactly which channels* MixMonitor is called on. By: Vadim (cron333) 2016-02-25 10:33:43.970-0600 Added debug_log_audio.txt. 1) 6501 calls to 79893839553 2) 79893839553 - answers 3) 6501 presses Hold 4) 6501 calls to 79097878265 5) 79097878265 - answers 6) 6501 makes "Attended Transfer" (on Zoiper-phone (REFER)) By: Rusty Newton (rnewton) 2016-03-13 17:05:57.021-0500 Closing this out as it does appear to be a duplicate of ASTERISK-25674. Please read through the comments on that issue for details on how to work with MixMonitor. | ||||