[asterisk-users] force soft hangup

[Tilghman Lesher]

On Tuesday 25 March 2008 10:17:54 Vieri wrote:
> --- Steve Davies <davies147 at gmail.com> wrote:
> > Using rtptimeout and rtpholdtimeout settings in
> > sip.conf
>
> I set
> rtptimeout=10
> rtpholdtimeout=30
> (just for testing; I know these values are way too
> low)
> then did a
> CLI> sip reload
> and waited more than 30 seconds.
>
> The SIP channel is still there (InUse).
> Channel              Location             State
> Application(Data)
> SIP/6010-b38d53e0    s at macro-dial:8       Up
> Dial(SIP/4053||tTwW)
>
> Should I interpret the above that it's in an infinite
> loop trying to dial/reach SIP/4053?

Given that you didn't give Dial a timeout, yes, it will try
forever, until it receives a response.  Note that this has
nothing to do with rtptimeout, as that takes effect when
the call is established, and the RTP packets stop flowing.
Without using a firewall rule between the two hosts, it is
somewhat difficult to mock up that situation (as the RTP
is still flowing, even if the audio is silent).

-- 
Tilghman