> Yes, this would be why I said that it is Asterisk's fault and provided > possible > workarounds. > > Thank you for your helpful and constructive criticism. LOL yes you could expect now everyone to be critical about something like this. Asterisk has been around for quite some time now (6+ years) and this sounds like a pretty basic problem that could cause a lot of failed calls with some SIP MTAs.
I would expect this kind of problem from an Asterisk version before 1.0.0 Martin _______________________________________________ -- 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
