[Asterisk-Users] sip qualify=yes interval

[Damon Estep]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of BJ Weschke
> Sent: Friday, January 27, 2006 6:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] sip qualify=yes interval
> 
> On 1/27/06, Damon Estep <damon at suburbanbroadband.net> wrote:
> > In an earlier thread Andrew Kohlsmith enlightened me on the use of
> > qualify in sip.conf to deal with a peer that is down.
> >
> > Since then I have been searching for information on how the behavior
of
> > qualify can be tuned.
> >
> > The wiki is vague on this;
> >
> > " Syntax:
> >
> >  qualify=xxx|no|yes
> >
> > where XXX is the number of milliseconds used. If yes the default
timeout
> > is used, 2 seconds.
> >
> > If you turn on qualify in the configuration of a SIP device in
sip.conf,
> > Asterisk will send a SIP OPTIONS command regularly to check that the
> > device is still online. If the device does not answer within the
> > configured (or default) period (in ms) Asterisk considers the device
> > off-line for future calls. "
> >
> > So;
> > qualify=1000|yes
> > means query for SIP OPTIONS, then take then unregister the peer if
no
> > response in 1000ms.
> >
> > But, how do you set/determine the frequency at which a peer is
queried?
> > Does this go on indefinitely after a peer fails to respond to make
sure
> > the peer is re-registered when available again? Can the interval be
set
> > on a per peer basis?
> >
> > Any documentation on this that you can point me to?
> 
>  It should actually be qualify=1000 if you'd like for the peer to be
> made unavailable when we don't get a response to SIP OPTIONs within
> 1000ms (1 second).

Figured that out, thanks.
> 
>  If the host is reachable, the next SIP OPTION attempt will not come
> until 60 seconds later. If the host isn't reachable, it will proceed
> to schedule SIP OPTION attempts every 10 seconds.
> 
>  These are defined constants in chan_sip.c

Seems silly to make these constant, I can think of many situations where
you might want to change them (heavily loaded system, many, many peers),
but I assume the sip options exchange is only a few packets... easy
enough to change the constants in chan_sip.c I suppose.
> 
Thanks for the info!