Hi everybody in new versions of Asterisk the RTP on SIP pass only througt the Asterisk, not directly between the endpoints like olders versions.
What happened whit this feature? (reinvite) Can you help me? _______________________________________________ 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
