[Asterisk-Dev] Problem with hanging up a SIP channel

[Marc Haisenko]

On Thursday 24 November 2005 20:15, imran ahmed wrote:
> Just a note that you cannot hangup the original incoming channel using
> ast_hangup (use ast_softhangup only on that channel) because asterisk
> expects that channel to be returned back to the dialplan.

Yes, I suspected that already as my tests showed ast_softhangup works on the 
original channel. But thanks for the hint anyway :-)
C'ya,
	Marc
-- 
Marc Haisenko
Comdasys AG

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