[cisco-voip] Adhoc conference calls drop without MTP

[Mooney, Nicholas]

Hi

 

That command didn't seem to make a difference, however I created a SIP
trunk from CUCM to the 2821 router and then from the 2821 to the GSM
gateway. Now it's SIP from CUCM through to the GSM. All the ringback
works fine, consultative transfer and Adhoc conferences are also good
(no disconnects). Also no need to enable MTP on the H.323 gateway so
there are no annoying messages about video bandwidth on PSTN calls.
There is however MTP on the SIP trunk in CUCM.

 

Although the original problem isn't fixed, I'm pretty sure you are on
the right track with it being a H.323 compatibility issue but taking out
H.323 seems to have solved it.

 

The reason for wanting to send the calls to 2821 first was that dial
peer hunting on the 2821 will allow calls to out via my ISDN if the SIP
trunk to the GSM gateway is full or not answering. I could send the
calls direct to GSM gateway from CUCM but I don't know to make CUCM hunt
to a h.323 gateway if a SIP trunk is not available (you can't add SIP
trunks to route groups can you?)

 

Thanks,

 

Nick.

 

From: Adam Frankel [mailto:afrankel at cisco.com] 
Sent: Friday, 7 August 2009 10:19 AM
To: Mooney, Nicholas
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Adhoc conference calls drop without MTP

 

More than likely the H323 gateway is not supporting emptycapabilities.
Make sure you are on a recent IOS version and put the
'emptycapabilities' command under voice service voip->h323.   Another
option would be to use an MTP which supports codec passthrough.  Keep in
mind with the MTP in use, you will need enough video locations between
the Video enabled IP Phone and the MTP as well as the MTP and the H323
gateway.  

Adam

-------- Original Message  --------
Subject: [cisco-voip] Adhoc conference calls drop without MTP
From: Mooney, Nicholas <Nicholas.Mooney at astrazeneca.com>
<mailto:Nicholas.Mooney at astrazeneca.com> 
To: cisco-voip at puck.nether.net
Date: 8/6/09 6:26 PM



Hi

 

I have an AudioCodes GSM (SIP) gateway connected to a 2821 H.323 gateway
via SIP which is then connected to CUCUM via h.323

 

Regular outbound calls work fine. CUCM sends all 0.@ calls to the H.323
gateway and then the gateway matches mobile phones call to a SIP
dial-peer and sends the call to the AudioCodes GSM gateway.

 

The only problem is when a call via the GSM gateway gets added to an
adhoc conference, it gets dropped/disconnected all together after about
10 seconds. If I configure the H.323 gateway to require an MTP then the
calls don't get dropped, however any CUVA (Video Advantage) user who
makes a PSTN call gets a message on their phone saying "Video Bandwidth
Unavailable". The CUVA doco says this is because an MTP is being used
and MTP's don't support video.

 

The only MTP's in my network are the default ones configured in CUCUM.

 

Any ideas on how I can get the calls via the GSM gateway not to drop
when added to an adhoc conference without enabling an MTP on the gateway
in CUCM? 

 

Or is there a way to have PSTN calls for CUVA users not show the "Video
Bandwidth Unavailable" message when it's clearly an audio-only call when
an MTP is enabled for the gateway? 

 

Thanks,

 

Nick.

 


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