Hello there, I'm using a Debian box with Asterisk 1.8.13.1 as a DID-PSTN gateway, so I can receive calls in a DID number and redirect it to my mobile line.
It has been working flawlessly for a few months, but I have noticed that some calls were not cut after losing one leg (the one with the DID server), and kept the PSTN leg active until I restarted the server (with the unexpected cost involved in the PSTN call). The relevant extensions.conf line is: exten => 34911234567,1,Dial(SIP/pstn/447123456789) And both DID and PSTN sip accounts have canreinvite=yes, so they can have direct media. I haven't collected any debug log nor any other relevant information. Does anybody know why something like this happens, or how can I cut a call that unexpectedly losed a leg? Thanks! L. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
