[asterisk-users] force soft hangup

[Vieri]

--- Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:

> > 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.

I've just run into another similar case.

This time a softphone (4012) which isn't even
registered anymore is still locking up a voice channel
through SIP/205 which is an FXO gateway connected to
PSTN. If I chanspy then I hear nothing/ total silence
(no tone plays).

Shouldn't rtptimeout do its job here and disconnect?
How can I make sure there really is no RTP flow?

Channel              Location             State  
Application(Data)
SIP/205-0a778a58     (None)               Up     
Bridged Call(SIP/4012-b3291db8
SIP/4012-b3291db8    s at macro-dialout-trun Up     
Dial(SIP/205/0666558844|300|TW

The FXO Grandstream gateway does not have Silence
Suppression enabled.

Thanks for any advice.



      ____________________________________________________________________________________
Never miss a thing.  Make Yahoo your home page. 
http://www.yahoo.com/r/hs