[asterisk-dev] SIP Not Showing Disconnect

[Paul Hewlett]

On Sunday 22 July 2007 21:20, Nicholas Blasgen wrote:
> I've been having some problems recently with Asterisk thinking a SIP phone
> is still connected.  These are GrandStream Budge Tone 102's and even after
> someone hangs up the AGI script I have running for them is still looping.

FWIW, we set up bt102's as not needing registration i.e. we set host=<actual 
ip> instead of dynamic and configured the phone to allow calls without 
registration and all our problems went away. Before that, the bt102's would 
randomly lose connection with asterisk (V1.2 BTW)

Paul
 
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