> Cassius Smith wrote: >> Hello all, >> I'm in final testing stages and preparing training for a new Asterisk >> rollout. I'm replacing a Cisco Call Manager system, and re-flashing >> the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, >> I use the "Confrn" softkey to invite other participants. I can add one >> other participant > > I think the phone itself is limited to 3 calls per line. > > Doug
It's got nothing to do with SIP. The phone itself is doing the audio mixing, and the phone is limiting you to 2 call legs (plus you, making a 3-way call). You don't usually see phones that will bridge more than 2 call legs because the required technical resources generally exceed those of the endpoint. Some 'better' phones such as the Polycom Soundpoint IP 6xx series phones will mix as many as 3 call legs (plus you, making a 4-way call). Still, the phone itself is bearing the burden of the audio mixing which explains the limitation. To conference more than 4 parties, you need a conference bridge (such as MeetMe). However, you can get clever with Asterisk and allow your end-user to transfer calls directly to a MeetMe conference with a single keypress! You can enable your end-user to collect conference participants one-by-one, rather than burdening your called parties with the need to dial in to a special phone number with a special access code! -Karl -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
