2009/10/16 Richard Kenner <ken...@gnat.com>: > I sent a query about this before, but have some further information and am > hoping somebody has a suggestion as to what to try next to debug this. > > I'm using an Asterisk box primarily for MeetMe conferencing. There are > two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works > fine between TDM channels. But when a SIP phone calls the conference, > there's no voice path *to* the conference. It can hear the conference > and its indicator changes appropriated from "not talking" to "talking", > but nothing from it gets bridged into the conference (the entering and > leaving tones work fine). > > Calls from the SIP phone to a TDM are fine. I tried the experiment of > having the SIP phone dial across the T1 to the PBX which will then tandem > the call back to Asterisk. When I do that, I have sound just fine. > "core show channel" look the same for both the Dahdi and SIP channels. > > This is very frustrating. Does anybody have any ideas? >
As a complete guess, I would check what codecs you are using on the SIP phone, and what transcoding paths are possible, particularly if you are using licensed codecs. Regards, Steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
