Hi

I have can get my phones to register with SER and dialout for PSTN via
my Asterisk box over a SIP channel to my VoIP provider. If the phone
requests hangup then the bridged channel on Asterisk gets destroyed
however if the called party hangups the channel stays up and the phone
connected. Anybody got any ideas?

Regards

Jon
-- 
Jon Farmer
Telford, Shropshire, UK
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