[asterisk-dev] Cisco 79x1 SIP Messaging

Robert Huddleston rhuddleston at gmail.com
Fri May 13 10:37:37 CDT 2011


On my 7941 I use conference button all the time. Running SIP firmware
against Asterisk.  Bandwidth takes a serious hit when I do it - so try to
avoid as much as possible

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Matthew Hoskins
Sent: Friday, May 13, 2011 11:20 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Cisco 79x1 SIP Messaging


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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