Hi Gurus, I we seen references to 'codec pass through feature' in the mailing list. SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand this feature, or point me to some examples etc.
Appreciate any pointers here. Thanks a bunch Sathya _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
