[Asterisk-Users] premature call release - SIP 480

[Olle E. Johansson]

Damon Estep wrote:
> When executing: Dial (SIP/##########@###.###.###.###,60
> <mailto:SIP/##########@###.###.###.###,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 ?

/O