[Asterisk-Users] MTP required for CCM integration ?

[Patrick Zwahlen]

Thanks for this one, Greg ! 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Greg
Oliver
Sent: mardi, 1. novembre 2005 16:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MTP required for CCM integration ?

You will probably also need to change the media exchange timers in CCM
if you are going to use it as a PRI gateway - otherwise asterisk -> 323
-> CCM -> PSTN calls will get dropped after 4 secs of ringing.


On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote:
> Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP.

> I will continue my tests, and maybe give a try to the patch you 
> mentionned. However, this will probably be too "cutting edge" for the 
> project ;-) I have a few questions, though:
> 
> - You mention that Cisco indicates that any H323 trunk with advanced 
> features needs an MTP. Can you point me to the place where you found 
> this ? Because as far as I can tell, this is not true for a trunk to a

> Cisco gateway.
> 
> - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, 
> I should have better luck with the Sourceforge version...
> 
> - From your experience, do you feel that a clean CCM<->* integration 
> is possible ? I am currently interested in simple feature (MoH, 
> transfers, maybe Call Park). A friend of mine is working on the 
> voicemail (unity) replacement/integration.
> 
> Thanks again for you quick support, and sorry for my late answer !
> 
> BR, - Patrick -
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan 
> Austin
> Sent: vendredi, 21. octobre 2005 18:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] MTP required for CCM integration ?
> 
> 
> > Is it required to use an MTP on the Cisco callmanager, when
> integrating
> > with asterisk (using h323) ?
> As of CCM 4.X, Cisco indicates that any H.323 trunk that will support 
> MoH/Transfer/etc need MTP resources.  Annoying.
> 
> > I am working on a project where the goal is to interconnect Cisco 
> > Callmanager (version 4) clouds together, using either SIP or IAX
> between
> > multiple * servers. Basic setup will be:
> 
> > PHONE - sccp - CCM (V4) - h323 - ASTERISK - iax - ASTERISK - h323 
> > -CCM
> > - sccp - PHONE
> 
> > I am working on the first half of it, which is:
> 
> > 7920 --- SCCP --- CALLMANAGER (V4) --- chan_oh323 --- ASTERISK 1.0.9
> 
> > I want to avoid the use of a gatekeeper.
> 
> > In that configuration, I am trying to get call transfer working. The

> > phone can call the DEMO app on asterisk, but then I cannot transfer
> the
> > call to another Cisco phone (on the same callmanager). I have some
> PCAP
> > traces if required. Basically, the 2nd phone rings, but there is no 
> > audio channel. After about 10 seconds, I see that that chan_oh323
> hangs
> > up the call.
> Sure will drop the call.  MTP does solve this.
> 
> > The idea was to avoid RTP streams through the call manager.
> Good plan, and one that I would consider a must for scalability and 
> quality.
> 
> > Now, if I define a Media Termination Point (MTP) on the Callmanager,

> > things work much better.
> 
> > I have also tried the new ooh323 with 1.2.0-beta1, but I couldn't 
> > get audio at all.
> Odd, I am using ooh323c.  I have a special test release, but the fixes

> for our CCM4 enviroment were added to CVS.  Are you using ooh323c from

> Asterisk-Addons or a download from Open Systems?
> 
> > I have read a lot about people having success with integratin CCM
> and*,
> > but without any details, especially around MTP configuration.
> 
> 
> > Any help would be greatly appreciated. BR, - Patrick -
> 
> http://bugs.digium.com/view.php?id=5374 has a patch that allows * to 
> send RTP packets when it is not receiving them.  I wasn't expecting 
> this result, but applying this patch resolved the disconnect when a 
> SCCP phone put a call on hold and allows transfers.
> 
> The bug/patch got quite a bit of early attention, but seems to have 
> languished.  Try it out and provide feedback.  Maybe enough success 
> reports will help get it rolling again.
> 
> Dan
> 
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