asterisk users - Jun 2006 - Bridging two existing calls (MeetMe, Sip Reinvite)

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.

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@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.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>

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@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.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
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