[Asterisk-Users] MTP required for CCM integration ?

[Dan Austin]

Comments inline

________________________________

From: asterisk-users-bounces at lists.digium.com on behalf of Patrick Zwahlen
Sent: Mon 10/31/2005 5:41 AM
To: Dan Austin
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MTP required for CCM integration ?



> 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.

Cisco introduced this requirement when 4.0 was released.  I have only
found it documented in the 4.X release notes.  As far as the H323 trunk
to the Cisco gateways, well I suspect Cisco has a way of handling that.
I prefer not to use MTP resources.  The Async patch solves the only
issue I had with ANY of the trunking methods betweek CCM and *,
which was disconnects during transfer/hold without the MTP.

> - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, I
> should have better luck with the Sourceforge version...

The ooh323c mailling list just had an announcement for a new release,
but the * channel driver has lagged a bit and needs to be updated.

> - 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.

I would say yes.  I am using * for services and not PBX functions. I
can get calls into * from SCCP phones and our H323 gateways.

> Thanks again for you quick support, and sorry for my late answer !


No problem, I hope it helps.
Dan
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto: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 <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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