> Damon Estep wrote: > > When executing: Dial (SIP/[EMAIL PROTECTED],60 > > <mailto:SIP/[EMAIL PROTECTED],60>) I get about 15 seconds of > > ringing, the called party rings, but if not answered in the ~15 seconds > > I get back SIP 480 temporarily unavailable. > > > > > > > > If the call is answered everything is fine and the call will continue as > > expected. > > > > > > > > The call is being passed to a TNT media gateway then to the PSTN via a > PRI > > > > > > > > The TNT reports Q850 cause 19 and responds with SIP 480 > > > > > > > > Somehow the TNT thinks the called stopped progressing on the PRI after > > 15 to 20 seconds. > > > > > > > > The Telco says they have done a capture and are getting a normal > > release, in other words their switch is not terminating the call or > > sending any Q850 message. > > > > > > > > I can not find any timers in the TNT that might cause this, and it is > > not reporting any expired timers. > > > > > > > > Any ideas? > > > > > > > > Does the SIP INVITE from * to the TNT contain a timeout? If so is it > > possible the, 60 in the dial command is being ignored? > > > > > > > > Either; > > > > > > > > The TNT got a maximum time parameter from asterisk and it has been > > exceeded, so the TNT responds 480, or; > > > > The TNT has a timer that expires after n seconds and sends the 480 on > > its own, or; > > > > The Telco is not seeing the progress they want to see and is sending the > > Q850 cause 19. > > > > > > > > Any opinions, suggestions? > > > > > Do you have qualify= on ? > This ended up being a global dial out timer on the media gateway (MAX TNT) in the SYSTEM profile. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users