[Asterisk-Users] Asterisk 1.2.0 fails to hang up using SIP

[Don Fanning]

Greetings all,

I just installed the beta now my SIP phone doesn't correctly hang up and
clear trunks once the call is answered.

First one is fine because it didn't answer on the far end.  The second
one stayed connected.

    -- Executing SetCallerID("SIP/100-5ab0", "<516301xxxx>") in new
stack
    -- Executing Dial("SIP/100-5ab0",
"IAX2/xxxxxxx at voipbuster2/0044289xxxxxxx") in new stack
    -- Called xxxxxxx at voipbuster2/0044289xxxxxxx
    -- Call accepted by 213.61.187.147 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/voipbuster2-8 is making progress passing it to SIP/100-5ab0
    -- Hungup 'IAX2/voipbuster2-8'
  == Spawn extension (internalselections, 01144289xxxxxxx, 2) exited
non-zero on 'SIP/100-5ab0'
    -- Executing SetCallerID("SIP/100-4740", "<516301xxxx>") in new
stack
    -- Executing Dial("SIP/100-4740",
"IAX2/xxxxxxx at voipbuster2/0044289xxxxxxx") in new stack
    -- Called xxxxxxx at voipbuster2/0044289xxxxxxx
    -- Call accepted by 213.61.187.147 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/voipbuster2-1 is making progress passing it to SIP/100-4740
    -- IAX2/voipbuster2-1 answered SIP/100-4740

*CLI> stop now
Beginning asterisk shutdown....
    -- Hungup 'IAX2/voipbuster2-1'
  == Spawn extension (internalselections, 01144289xxxxxxx, 2) exited
non-zero on 'SIP/100-4740'
Executing last minute cleanups
  == Destroying musiconhold processes
Yuck! Error in buffer handling...: Connection reset by peer
Asterisk cleanly ending (0).
subspace:/etc/asterisk#