ConfBridge over trunk with follow_talker

I’m working on a university project and trying to set a confbridge with video_mode = follow_talker with users on two Asterisk PBX.

Here’s the background information: lets assume that a company is situated in two buildings, each has an Asterisk PBX and own users connecting to its PBX with SIP protocol. The Asterisk PBXs’ are connected with a IAX2 trunk. There is a need of creating a video conference with users from both buildings.

So I’ve set up a ConfBridge on Asterisk1 with follow_talker video setting, and created an outgoing calling rule on Asterisk2. This worked almost perfect. Every user receives video. The only issue is that users from Asterisk2 are never set as video source, even if other users are muted. I’m confused what to do.

My only ideas are:

  1. Should Asterisk2 be sending some AMI notifications to Asterisk1?
  2. Should a seperate confbridge be created on Asterisk2 and somehow linked to confbridge on Asterisk1?
  3. Should the trunk between Asterisk1 and Asterisk2 be set up on protocol SIP rather than IAX2?

Please help! I’ve got 2 days left and I’m stuck…

I’ve found a workaround. The problem is that IAX2 trunk cannot handle ConfBridge with video support.

When a user from other Asterisk PBX joins the ConfBridge conference without uploading video stream, its working correctly. Everybody is hearing him and he gets the video. But when that user starts uploading his own video stream, it works for 3 seconds and then he gets disconnected. The following warnings are shown:

chan_iax2.c: Received mini frame before first full video frame ; about a 1000 times
chan_iax2.c: Can’t compress subclass 2097217

So I’ve switched the trunk protocol from IAX2 to SIP and it works perfectly now. Should I report this as a but, or I’m missing something with IAX configuration?


Ick. You’re welcome to report it, but it’s unlikely to get serious attention - there’s an easy fix, as you note.

Thanks for your reply Malcolm.

Which fix did you mean? The only one I have found is here:

But this fix was added 2 years ago, and I have compiled the newest Asterisk version from trunk branch.

Are you sure there’s any other easy fix for this issue?


By fix, I mean workaround, sorry - use SIP.