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

[Dan Austin]

> If using CCM >= 4.0, using SIP trunks will alleviate a lot of
headaches.
True, if you don't need any codec other that G711.  According to
Cisco's docs (which could be wrong), SIP trunking in 4.X only
supports G711.

In any case I wanted the archives to have correct information
about CCM/H323/MTP and Asterisk since any details in there from
we in the past were likely very wrong.

Dan
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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