On Fri, Oct 27, 2006 at 08:10:30AM -0400, Steven wrote: > If you are calling from a SIP phone through asterisk and through a Digium > card, one could argue that the Digium card IS farside of > the SIP phone. > > SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN -- > Destination.
If you're calling from a SIP phone through Asterisk to a digital PRI card to the PSTN, then your system has no hybrid. Except, potentally, the SIP handset itself. You may be required to cancel echo generated by other people's systems. E.g: somewhere on the PSTN. > > I would argue that the Digium card IS on the farside of asterisk as > far as the SIP phone is concerned. > > We just switched from Legacy PBX to Asterisk and we get occasional echo. > Everything past the Digium card is the same as the old PBX. > We never got echo on the old PBX. > How do you connect to the PSTN? Digital or analog? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users