[asterisk-users] dead sip channel

[Jerry Geis]

I have ran into a case using 1.4.22 where a SIP call to an asterisk 
client (running a slow PC) to ALSA
does not hangup the call when it is done. The server is using call files 
to initiate the call, the client answers on
the ALSA port, the server plays the message and hangs up.

I found that SOMETIMES -its hard to recreate - that the slow pc keeps 
the SIP channel active. further calls in
are getting a busy signal and the one call is NEVER hung up.

How can I detect this and hang up the channel on the slow PC. I verified 
on the server that it thinks NO calls are active.

my context looks like this:
[mycontext]
exten => s,1,ChanIsAvail(Console/Dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup

[smvoice-busy]
exten => s,1,playtones(busy)
exten => s,1,wait(10)
exten => s,1,Hangup

Thanks,

Jerry