[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)

[Vieri]

--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:

> Vieri wrote:
> > Did you try a "show channels" to see if there were
> > stale channels for peer 200?
> >
> > I had the same problem you describe but it was due
> to
> > "hung" channels (used * 1.4.18.1 with rtp*timeout
> and
> > saw "inuse" peers during the pre-timeout periods
> even
> > though the agents weren't on a call).
> >   
> No, i don't , but how do do you fix this problem ?
> with rtp timeout ?

rtp*timeout for sip peers is not a fix but a
workaround.
Try to set both values and reload sip.
Then when you witness what you posted try doing a
"core show channels". You can then try to "soft
hangup" a stuck channel or wait for the rtp*timeouts.



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