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. 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. -- -- Steven http://www.glimasoutheast.org "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > On Thu, Oct 26, 2006 at 04:17:16PM -0700, Alyed Tzompa wrote: >> >> Echo is generated by the analog end to where you place the call, not the IP >> side of it. >> >> As far as I know the echo cancelation in the Asterisk can only be tweaked in >> the zapata.conf (since IP calls don't generate it) >> >> I'm afraid there is little you can do to here. > > A digital zaptel card (PRI/BRI) does not generate them either. > > If the echo would be generated on your side, you wouldn't be the one to > hear it. > > -- > 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 > _______________________________________________ --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
