You can use H.323 on Asterisk and setup CM to use an H.323 gateway to
Asterisk, or setup a gatekeeper and have both ends talk to the
gatekeeper. I have redundant CM boxes, so I HAD to use a gatekeeper and
set it in proxy mode because I had media path issues (the call initiated
from one box, but for some reason the media path on outside calls came
from the other box and we had one way audio). If there is just one CM
box, then you probably don't need a gatekeeper.
[EMAIL PROTECTED] wrote:
I need to pick all the Asterisk and Cisco People a little.
My company has a Cisco Call Manager 3.3, configured via h323 gateways. I
have remote users that I want to place a SIP Server on the external WAN
and be able to connect their phones to the system and be able to get calls
and call people in the office going through the Cisco Call Manager and the
h323 router. My only problem is that Cisco Call Manager 3.3 does not
support sip trunking. Is there anyway this can be done.
Please shed some light on this topic.
Thanks.
Goran
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