[asterisk-dev] Cisco 79x1 SIP Messaging

[Gareth Palmer]

On Thu, 2011-05-12 at 13:01 -0700, Dan Austin wrote:
> Matt wrote:
> > Server side conferencing looks like it's going to be a little more difficult, as the call > legs are within the XML dialog.  Perhaps this would be a good time to instantiate an XML > parser?  What's the desired actions from an asterisk perspective?  I'm assuming it would > go something like this:
> 
> > 1. User establishes Leg A and presses Cnfrnce
> > 2. User establishes Leg B and presses Cnfrnce to join the calls.
> > 3. Phone sends a SIP dialog with the two callid's embedded within an XML dialog
> > 4. Asterisk creates a meetme instance and joins the three channels.
> 
> > I'm sure there are a ton of steps that go into step 4. :)
> Step 4 strikes me as exactly what the new bridging API was meant for.
> A few caveats spring to mind-
> 	1.  I believe Cisco uses the phone to mix the audio when-
> 		A.  Only three parties are involved
> 		B.  Direct media is enabled and possible
> 		C.  The codecs are identical on both legs
> 	2.  It needs to be possible to cancel out after step 2
> 	3.  Each leg should be tracked allow to dropping a specific leg

struct ast_bridge keeps track of the current participants, just need to
iterate over bridge->channels.

> The first item might be specific to SCCP, and/or not worth the
> extra effort.

With the SIP firmware the phone does either one or the other of those
based on how the Confrn softkey is defined.

> Dan
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev