[Asterisk-Users] Cisco Call Manager and H323 trunk correction (MTP)

[Greg Oliver]

If using CCM >= 4.0, using SIP trunks will alleviate a lot of headaches.

On Tue, 2005-11-15 at 16:33 -0800, Dan Austin wrote:
> I posted a couple weeks back about our experiences with H323 trunks on
> CCM. 
> As of version 4.0, the Cisco documents state that a 3rd party H323
> gateway 
> requires a Media Termination Point..
> 
> At the time I said that I have Asterisk working with the ooH323c
> version  of 
> chan_h323 with out an MTP.  I just found that another engineer had
> been 
> twiddling with the CCM config, and we were using a MTP.
> 
> I retested chan_h323 without the MTP, and indeed per the Cisco docs, 
> when a phone connected to CCM puts a call placed through chan_h323 on 
> hold, the call is disconnected.  This IS NOT a bug with asterisk or
> the 
> chan_h323, but a known Cisco quirk.
> 
> Cisco's own H323 gateways are capable of dynamically
> creating/connecting 
> to a MTP.  Which permits calls to/through them to allow rtp re-invites
> and 
> still preserve a call during media transitions.
> 
> I thought I should post this for the archives in case anyone searching
> for 
> details about connecting CCM to Asterisk found my earlier
> misinformation.
> 
> Dan
> 
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