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List:       asterisk-dev
Subject:    Re: [asterisk-dev] SIP Channel does not release.
From:       Luigi Rizzo <rizzo () icir ! org>
Date:       2006-10-31 23:28:48
Message-ID: 20061031152848.A18031 () xorpc ! icir ! org
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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@transit:1  Down    AppDial((Outgoing Line))
> SIP/192.168.1.35-082 935055490@transit:1  Ring    Dial(SIP/935055490@mediagw1)
> 
> with IAX2 this does not happen.
> 
> Regards,
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