Hi,
I am new with asterisk and everything that deals with. 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,
azt
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
