Let me ask a simple question for some to answer. I need asterisk toIIRC the threewaycalling option only applies to analog extensions..
perform three-party-conference between SIP phones. Is there any setting
required on sip.conf ? I see Zapata.conf has threewaycalling=yes
parameter. Is there anything similar required in sip.conf or elsewhere.
I do not use any hardware. I use * with Cisco ATA 186 adapters. Thank
you in advance.
If you want it on SIP then you simply have to make sure the phone supports it.. Both my Grandstream and Snom support it so I am sure most phones do.. in fact off the top of my head the only one I know that specifically does not support it is X-Lite..
Later..
_______________________________________________ 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
