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. Thanks! -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- 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
