[Asterisk-Users] Re: [asterisk-dev] Bridging two existing calls (MeetMe, Sip Reinvite)

[Matt Florell]

Hello,

I developed a patch to do bridging of two active channels over a year
ago and have been using it in production ever since then. I was
promised that it would make it into 1.2 at the time, but clearly that
didn't happen.
http://bugs.digium.com/view.php?id=4297

I gave up trying to push it after 6 months and heath1444 took up the
cause and created a new patch:
http://bugs.digium.com/view.php?id=5841

Not sure but supposedly there is a similar feature being planned for
1.4. We'll see if it actually happens.

I don't know how this works with SIP reinvites, you will have to try
it out and let us know.

MATT---

On 6/16/06, Matt King <m at orderlysoftware.com> wrote:
> 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.
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