[asterisk-dev] SIP Phone Conference from Asterisk.

[Wai Wu]

I still have no idea what the original poster wants to do. Can he just
do this

SIP A calls SIP B, and transfer SIP B  to an extension into meetme
SIP A calls SIP C, and transfer SIP C  to the same extension into meetme
SIP A calls that same extension into meetme 

>From his original post, how is asterisk going know the intend of SIP A?

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Kaloyan
Kovachev
Sent: Tuesday, April 04, 2006 10:36 AM
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] SIP Phone Conference from Asterisk.

On Tue, 4 Apr 2006 09:41:12 -0400, Patrick Greene wrote

> We ran into this exact scenario about a year ago and wanted to be able

> to do
what you 
> want to do, but gave up.
> 

 it is not impossible, i think. I am not C programmer and may be
completely wrong, but it should be easy to modify the attended transfer
feature to forward all 3 parties to a predefined (in features.conf or
channel variable) extension, which will start MeetMe or other conference
app. If instead of 'ast_bridge_call_thread_launch(tobj);' something
similar to G option of Dial is made, when the transferer uses disconnect
feature code, then instead of hangup to forward the channel to the same
extension, but diferent priority to separate the transferer.
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