Asterisk group video calling

Asterisk Asterisk SIP
1

Hi All,
I am trying to call from sip door entry panel to multiple sip monitors.
If I dial 1 extension from door panel to monitor every thing works fine. I get the video before answering the call.
But when I dial two extensions (using ring group) then I dont get the video before answering the call.

My dial plan is as follows:
exten => 8001,1,NoOp(dialing)
exten => 8001,2,Dial(SIP/2001&SIP/2003)
exten => 8001,3,Hangup

My CLI output is as follows:
– Executing [8001@internal:1] NoOp(“SIP/2000-00000000”, “dialing”) in new stack
– Executing [8001@internal:2] Dial(“SIP/2000-00000000”, “SIP/2001&SIP/2003”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/2001
– Called SIP/2003
– SIP/2001-00000001 is ringing
– SIP/2001-00000001 is making progress passing it to SIP/2000-00000000
– SIP/2003-00000002 is ringing
– SIP/2003-00000002 is making progress passing it to SIP/2000-00000000
> 0x7f63f40141e0 – Probation passed - setting RTP source address to 192.168.0.153:6000
– SIP/2001-00000001 answered SIP/2000-00000000
– Remotely bridging SIP/2000-00000000 and SIP/2001-00000001
== Spawn extension (internal, 8001, 2) exited non-zero on ‘SIP/2000-00000000’

Please can anyone suggest where I am doing wrong?
thanks in advance,

2

In that case Asterisk does not forward the media before answer, as there could be multiple streams.

3

thanks for your suggestion. Do you think we can forward multiple streams ?

4

There is no capability to do that currently, any support for that would require code changes. I don’t believe anyone has worked on doing so.

5

Well from the door panel I can also dial two extensions at a same time.In door panel there is a speed dial option where I can specify two extensions. Means instead of dialing the group extension 8001
I can specify SIP:2001@IP and SIP:2003@IP, In this case door panel dial two monitors simultaneously with video before answering the call. In that case my CLI gives the following output:

Executing [2003@internal:1] Dial(“SIP/2000-00000003”, “SIP/2003,60”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/2003
– Executing [2001@internal:1] Dial(“SIP/2000-00000005”, “SIP/2001,60”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/2001
– SIP/2001-00000006 is ringing
– SIP/2001-00000006 is making progress passing it to SIP/2000-00000005
– SIP/2003-00000004 is ringing
– SIP/2003-00000004 is making progress passing it to SIP/2000-00000003
– SIP/2003-00000004 answered SIP/2000-00000003
– Remotely bridging SIP/2000-00000003 and SIP/2003-00000004
== Spawn extension (internal, 2001, 1) exited non-zero on ‘SIP/2000-00000005’
== Spawn extension (internal, 2003, 1) exited non-zero on 'SIP/2000-00000003’
I am trying to replicate this scenario in my dialplan. As i dont want to dial two extensions from the door panel

6

I’m afraid you can’t if you want media to flow. It just doesn’t work that way.