[asterisk-dev] Cisco 79x1 SIP Messaging

[Matthew Hoskins]

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

What's the configuration needed to restore the Confrn button to do local mixing of the channels?  I'm currently not using TCP SIP with the 9.x phones because my conference buttons don't work (trying to do server side conferencing).


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
> 
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