Hi! > If SIP/U2 transferred the call to an extension that made use of the > "switch statement"... What would the call path be? > > Would the call traffic go from A1 in A2 back out of A2 to A3? > ...or would it be "switched" and go directly from A1 to A3?
The theory - as far as I was able to find out - involves: transfer=yes/no in iax.conf canreinvite=yes/no in sip.conf Next to that you might have codec issues involved, i.e. if the server A1 has a different set of allowed codecs than A2 and A3. I am not sure if incompatible codecs result in a) an aborted call or b) a routed call even if transfer=yes. Cheers, Philipp _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
