Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. The system is already able to make and receive calls from the PSTN, although, only after the call has been established it can be hung up with success; when it is still ringing, if any side hungs up the call, it still keeps ringing on the other side. Observing with Ethereal, we concluded that in this erroneous cases, the CANCEL SIP request isn't transmitted from the SER to Asterisk (if cancelled from the VoIP side) being transmitted a "404 User Not Found" message from SER to Sip Phone. If hung from the PSTN side, the sip phone keeps calling after that, and ends calling by time-out being observed a "486 Busy Here" status message from Asterisk to SER and then from SER to sip phone. Any help, please? Regards, Ricardo. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers