You are usinfg sip or iax ? Its possible to prevent in both cases for sip under peer definition you can put canreinvite=no and in iax2 you can put transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for this on voip-info.org wiki for more info .
On Jan 25, 2008 7:03 PM, <[EMAIL PROTECTED]> wrote: > I have a call coming in from Asterisk-A going to Asterisk-B where it's > determined that the called party is in fact yet another number in Asterisk-A > so a new call is created from B to A and the two calls bridged (by Asterisk) > at Asterisk-B. > > > > Originating Caller ==> Asterisk-A ==> Asterisk-B ==> Asterisk-A > > > > Now, what happens is that in my case both A and B are on the same network > and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B > out and the original caller talks directly to the extension hosted in > Asterisk-A without the call path going the round-trip to Asterisk-B. > > > > Is it possible to prevent this optimization from happening? Any way to > control if it happens at all, or can it be selected on per-call basis > somehow? > > > > Can I find anywhere more details of call path optimization and it's > configuration, use, functionality and behaviour? > > > > tnx, > > Baldvin > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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