[asterisk-dev] Cisco 79x1 SIP Messaging

[Matthew Hoskins]

That patch looks great!  I'm glad I posted to the list first.  Any assistance I can provide, I'd be happy to.  I unfortunately don't have access to CUCM, but there was a response on the list from someone who may be able to help.

The CFwdALL, BLF's, and DND worked perfectly first time once I applied the patch! 

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. :)


Matt
----- Original Message -----
From: "Gareth Palmer" <gareth at acsdata.co.nz>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Wednesday, May 11, 2011 9:30:30 PM
Subject: Re: [asterisk-dev] Cisco 79x1 SIP Messaging

On Wed, 2011-05-11 at 20:37 -0500, Matthew Hoskins wrote:
> Hello,
> 
> I was wondering if anyone on the list was working on implementing the extended functionality of the cisco 79x1 SIP messaging?  It looks like cisco is implementing call parking, conferencing, dnd, pickup groups, call forwarding, etc via SIP REFER/INVITE messages with XML dialogs.  

I have been working on issue 13996 [1] that provides support for most of
that.

There is an additional, unreleased patch that supports receiving an out
of dialog REFER's for iDivert support as well.

> I have access to a 7945 and am willing to dig in and see if I can extend the sip codebase to support these functions.  I just wanted to check on the list first to make sure other people weren't working on it, or there wasn't a specific reason they hadn't been implemented.  

Access to the unified endpoints is not a problem. Lack of access to, or
packet captures from a full CUCM installation is. 

I was able to get required SIP+XML for DND and CFwdALL from a CME
instance but not BLF ringing indication or server side conferencing.

CME only appears to support a sub-set of the SIP endpoint's features.

> Thoughts???

[1] https://issues.asterisk.org/view.php?id=13996


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