[Asterisk-Users] * and Cisco Callmanager Interconnection

[Dan Austin]

I had it 'working' and quickly moved back to H.323 using
chan_oh323.  I had great expectations for SIP support in
CCM4, but ended up disappointed.  The requirement for a 
Media Termination Point to handle DTMF and support for only
the G.711 codecs was just too much.

I'm hoping that Cisco will continue to improve the CCM
SIP trunk and it will become a viable option, but at the
moment it is really only usefull as something to experiment
with.

Dan 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Parker,
Blake (MIS)
Sent: Wednesday, March 23, 2005 5:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] * and Cisco Callmanager Interconnection

Has anyone had any luck getting a SIP trunk up and working between
Callmanager and Asterisk?  If so were there any steps you had to take
that were not in the documentation on wiki?

Blake
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