[asterisk-dev] Bridging two existing calls (MeetMe, Sip Reinvite)

[Matt King]

Hello,

I know there's a problem with Asterisk at the moment in that while it's 
easy for one caller to dial another (using the dial command), it's 
tricky to connect two calls that are already in progress.

I've been using MeetMe to achieve this (with each caller's call being 
directed to a dynamically created conference room programatically), and 
this is working - kind of - but this results in a conference instead of 
a bridged call, so

    - we can't use the normal Dial parameters for transfer etc,
    - the other caller is not disconnected automatically when one party 
hangs up, and
    - (most importantly) we can't get SIP to reinvite.

The SIP reinvite issue results in increased bandwidth costs, extra 
latency/echo and reduced call quality when compared with Dial (as the 
media path has to include Asterisk with MeetMe, but not with Dial).

Does anybody know of any other way to bridge two existing calls with 
Asterisk, that will allow SIP to reinvite?

I've already asked on the IRC channel, searched the list archives and 
had a look through the bug tracker.  I'm cross-posting this to the dev 
list too as this my last resort before making a feature request/bug post...

Hope this helps,

Matt.