thr3ads.net - asterisk users - [Asterisk-Users] MTP required for CCM integration ? [Oct 2005]

If this information is useful, please help other people find it:
Share via:

Patrick Zwahlen

2005-Oct-21 07:38 UTC

[Asterisk-Users] MTP required for CCM integration ?

Hi,

Is it required to use an MTP on the Cisco callmanager, when integrating
with asterisk (using h323) ?

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.

The idea was to avoid RTP streams through the call manager.

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.

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 -

Dan Austin

2005-Oct-21 09:37 UTC

head link

[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

Patrick Zwahlen

2005-Oct-31 06:41 UTC

head link

[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.

- 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@lists.digium.com
[mailto:asterisk-users-bounces@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

Dan Austin

2005-Nov-01 12:18 UTC

head link

[Asterisk-Users] MTP required for CCM integration ?

Comments inline

________________________________

From: asterisk-users-bounces@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com
<mailto:asterisk-users-bounces@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

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>


-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/ms-tnef
Size: 8160 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20051101/76978c57/attachment.bin

Patrick Zwahlen

2005-Nov-02 03:59 UTC

head link

[Asterisk-Users] MTP required for CCM integration ?

Thanks for this one, Greg ! 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@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@lists.digium.com
> [mailto:asterisk-users-bounces@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
> 
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Patrick Zwahlen

2005-Nov-02 04:02 UTC

head link

[Asterisk-Users] MTP required for CCM integration ?

Hi Dan,

Your comments definitely help. Thanks a lot.

I'll probably have more remarks / questions early next week.

BR, - Patrick -
________________________________

From: Dan Austin [mailto:asterisk-users-bounces@lists.digium.com] On
Behalf Of Dan Austin
Sent: mardi, 1. novembre 2005 20:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MTP required for CCM integration ?


 
Comments inline

________________________________

From: asterisk-users-bounces@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com
<mailto:asterisk-users-bounces@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

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>

asterisk users - Oct 2005 - MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?

[Asterisk-Users] MTP required for CCM integration ?