[asterisk-users] force soft hangup

[Vieri]

--- Steve Davies <davies147 at gmail.com> wrote:

> On 25/03/2008, Gordon Henderson
> <gordon+asterisk at drogon.net> wrote:
> > On Tue, 25 Mar 2008, Vieri wrote:
> >
> >  > How can I "force" soft hangup (if that makes
> sense)?
> >  >
> >  > "show channels" reveals a stale sip channel.
> It's of
> >  > an analog phone behind a Grandstream ATA which
> was
> >  > communicating with another SIP softphone. The
> latter
> >  > crashed. A soft hangup of the softphone seems
> to have
> >  > worked but it doesn't for the analog/ATA phone.
> "show
> >  > hints" also shows that it's InUse. But of
> course it
> >  > isn't and noone can contact this extension
> since I
> >  > disabled call waiting for it (I also rebooted
> the ATA
> >  > and forced re-registration of the ATA SIP
> client).
> >  >
> >  > So how can I "kill" this channel without
> restarting
> >  > the asterisk daemon?
> >
> > Strangely enough the command in the CLI is:
> >
> >    soft hangup <channel>
> >
> >  Just type soft hangup then push the TAB key and
> it will auto complete for
> >  you ...
> 
> I think the OP's point is that "soft hangup ..."
> does not hang up one
> of the channels successfully. I have seen this
> before, usually from
> soft-phones that do not disconnect cleanly, or WiFi
> phones that lose
> signal during a call.
> 
> Using rtptimeout and rtpholdtimeout settings in
> sip.conf seems to
> improve matters, as does using more recent versions
> of asterisk - You
> do not say what version you are running.

Thanks Steve,

I'm using 1.2.26.2 and upgrading to 1.2.27 today.
rtp*timeout sounds like it could do the job (just hope
it won't do false hangups) and I think that these
settings are available for 1.2.



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