[Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper

[Joe S]

Hi,

I am new with asterisk. I was wondering if there is a way to call a OH323 
user or SIP user using Netmeeting/SJPhone with H323 as the default protocol 
without having a gatekeeper.

I can make a call from SIP to OH323 by specifying it in the extensions.conf 
file, like:

exten=>1001, 1, Dial(OH323/10.10.10.1)

so I was wondering if there was a way to call from OH323 to SIP or OH323.

Thanks I appreciate any thoughts and ideas,

joe