Summary:	ASTERISK-28344: Wrong music on hold handling on multi party attendant transfer
Reporter:	Parantido Julius De Rica (parantido)	Labels:	patch pjsip
Date Opened:	2019-03-22 02:58:47	Date Closed:	2019-04-16 12:09:43
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General Resources/res_musiconhold
Versions:	1.8.30.0 11.0.0 12.0.0 13.0.0 15.0.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Linux CentOS 7, Ubuntu Server 14.04.4 LTS	Attachments:	( 0) chan_sip.c.diff ( 1) channel.c.diff ( 2) channel.h.diff ( 3) res_musiconhold.c.diff
Description:	On a multi party attendant transfer scenario music on hold is not correctly handled on final bridged channels. - Scenario - 1) Secretary A place a call to Secretary B because Boss A needs to talk to Boss B. 2) Secretary A correctly talked with Secretary B and place channel on Hold to contact Boss A. 3) Secretary B correctly listen to music on hold. 4) Secretary B place channel with Secretary A on hold to place a call with Boss B. 5) Secretary B talk with Boss B and transfer it to Secretary A. 6) Boss B do not hear music on hold. This issue affect all asterisk release starting from 1.8.32.0 using both chan_sip than chan_pjsip. Multiple device are used to reproduce described scenario (snom, polycom, grandstream and so on).
Comments:	By: Asterisk Team (asteriskteam) 2019-03-22 02:58:48.640-0500 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Parantido Julius De Rica (parantido) 2019-03-22 03:22:53.898-0500 Attached you can find diff for asterisk 1.8.32.3 files (the one I used to apply patch). This will fix described issue By: Chris Savinovich (csavinovich) 2019-04-01 12:53:05.942-0500 Hello Parantido, thank you for submitting this issue, unfortunately because your scenario uses chan_sip we are going to have to refer you to community help for finding a solution to your issue. However, may I suggest that if you were to recreate the same scenario this time with all extensions and components using chan_pjsip we will be happy to support it.  If you chose to do so, and still find the issue persists, we would require you that this time you attach all related sections of all corresponding files: pjsip.conf, extensions.conf etc, so that we are able to recreate your environment. Thanks C. Savinovich By: Parantido Julius De Rica (parantido) 2019-04-01 13:06:28.067-0500 Hi Chris, I already attached patched files to solve issue. Chan PJSIP is also affected but I'm not really using it. Applying my patch to master code is up to you, let me know if interested in. Regards By: Chris Savinovich (csavinovich) 2019-04-01 13:35:13.448-0500 Hello Parantido,    In my opinion, if it is up to me, to approve your patch I would have first to test the error and to do that I would have to recreate it, and for that I would have to have the requested sections of the configuration files. However, I can also suggest if you instead chose to submit your patch via gerrit (https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage), you are welcome to do so and therefore with some of the reviewers more closely familiar with the code they will probably approve it without requesting to create a test. Thanks C. Savinovich. By: Richard Mudgett (rmudgett) 2019-04-02 07:55:26.204-0500 Please use "diff -u" or "svn diff" on all your patches. Patches which include alternate formatting are almost certainly going to be thrown out or ignored; there are too few hours in the day to wade through difficult-to-follow C code fixes without the help of diff -u. Thanks! By: Asterisk Team (asteriskteam) 2019-04-16 12:00:02.521-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines