Hi, How asterisk decides whether to do media relaying or not? For SIP I've found that "canreinvite=yes" allows me to use * only for signalling, RTP stream will flow between endpoints only. Are such things possible when calling from SCCP channel to SIP for example? SCCP to SCCP?
Thanks in advance! -- Alexei Chetroi _______________________________________________ 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
