[asterisk-users] force soft hangup

[Steve Davies]

On 25/03/2008, Tilghman Lesher <tilghman at mail.jeffandtilghman.com> wrote:
> 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).

I would have assumed that "Up" indicates that it thinks the call is
already in-progress, and that RTP should be flowing.

You will not be able to change the rtptimeout values of an established
call as AFAIK these settings are copied into the channel object when
it is created, and not updated if SIP is reloaded.

So... I think that channel is stuck until a restart, but you may be
able to reduce its occurrence in future by using rtptimeout, and
possibly also by using Dial timeouts as Tilghman pointed out.

I use rtptimeout as standard these days, and have not seen a channel
stuck-hard for quite a while (Nor have I seen false hang-ups).

Regards,
Steve