[asterisk-dev] SIP Channel does not release.

[Luigi Rizzo]

On Wed, Nov 01, 2006 at 02:48:00AM +0500, Anton wrote:
> 
> Hello,
> Just to point out to the SIP issues which exist in 1.4svn (up to SVN-branch-1.4-r46663) (and possible trunk). 
> When SIP call is made from one ASTERISK box to another - 
> and second box RINGING, if we hangup the caller station, 
> without answering callee, callee still rings. channel list gives that call is active on the callee asterisk, 
> while calling station has no calls in cahnnel list.
> Seems SIP does not send release.

i wonder if it isn't a problem with your dialplan that passes
the 'h' extension to the same destination ?

cheers
luigi

> SIP/mediagw1-0825cfb 935055490 at transit:1  Down    AppDial((Outgoing Line))
> SIP/192.168.1.35-082 935055490 at transit:1  Ring    Dial(SIP/935055490 at mediagw1)
> 
> with IAX2 this does not happen.
> 
> Regards,
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