Ok. I've noticed a thing: when you ring a sip phone, and hangup before it answer, asterisk sends a CANCEL to the phone to abort the current operation (in this case, the INVITE). and this's correct according to rfc.
But now... when a sip phone A is ringed from a phone B , and that call from B is picked up by the phone C via *8 , asterisk sends 'BYE' to the phone A ( C & B are bridged ok). But according to rfc, that's wrong, since 'BYE' must be sent to release an active call . The right thing to do is to send a CANCEL to A, since we want to abort the pending INVITE. I'm right ? That's a bug in asterisk ? I've found that using the budgetones phone. They'll go crazy if a INVITE is aborted by a BYE instead of a CANCEL. Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y ------END GEEK CODE BLOCK------ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users