On 14 Feb 2014, at 00:43, Kevin Harwell <[email protected]> wrote:
> Greetings, > > There is a bug currently in Asterisk that essentially boils down to MOH > not being [re]started for an attended transfered call when the > transferor was listening to it. > > For example, here is a specific scenario (issue ASTERISK-19499): Alice > calls Bob, Bob starts a SIP attended transfer into a confbridge that is > set to play music when empty. Alice hears music since she is now on > hold and Bob hears music from the confbridge. Bob then completes the > transfer and then Alice enters the confbridge, but hears no music. > > This situation occurs because during the transfer and channel masquerade > the application has no way to know the channel was swapped out. There > are a couple of ways to approach a fix for this: > > 1) Use a channel fixup in the applications that are affected. A small > change to the "do masquerade" function has to be made to get this to > work too. > > 2) In the relevant channel drivers when an attended transfer occurs > check to see if MOH is currently playing on the transferor. If so, once > the transfer occurs start MOH on the transferee. This idea was put > forth by Olle Johansson (currently implemented in his > rana-moh-queue-transfer-1.8 team branch). Albeit a few changes will > have to be made in order to use it in the main branches, but the idea > will essentially remain. > > I like option 2 as it seems to be a bit less intrusive (no changes to > masquerades and such). Also in 12+ the logic can be put into the bridge > transfer code reducing the changes further. > > Thoughts? Ideas? Something I have missed or need to be aware of? > > I'll start to move forward with #2 soon and put it up for review. > Comments and feedback always welcome. CHeck this branch: http://svnview.digium.com/svn/asterisk/team/oej/rana-moh-queue-transfer-1.8/README.rana-moh-queue-transfer.txt?revision=393703 /O -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
