[Asterisk-Users] Delays in sip invites.

Ivan Cetta icetta at gmail.com
Tue Nov 1 11:35:56 MST 2005


No. in any place, even i haven't any agi script in the dialplan.

exten => _87XX,1,Dial(SIP/${EXTEN},60)

This is the only i have to call the sip ata.

Also, something i forgot to mention, i have in the sip.conf and the
audiocodes the srvlookup disabled. Browsing the problem in the list i see
that someones have something similar and was dns problem. But in our case,
changing that didin't resolved it.

Ivan Cetta.


2005/11/1, Iqbal <iqbal at gigo.co.uk>:
>
> are you doing enum lookups anywhere , had a problem with enum lookups
>
> Iqbal
>
> Ivan Cetta wrote:
>
> > Hi all.
> >
> > We are having some strange behaviors in our asterisk box. A Dual p4
> > Xeon 3ghz, with 2gb of ram and a a digium TE410p card.
> >
> > The system is an Asterisk V 1.0.9 with Unicall 0.0.2c (Argentina
> > variant) to interconnect with an Alcatel pbx and to the pstn.
> >
> > It has an average of 10 concurrent calls with several queues with
> > statics agents all of them. (an average of 24 agents logued in sip
> > atas, an audiocodes mp-124)
> >
> > The problem is that Randomly we are having some delays when the
> > application dial is executed in the dialplan, trying to call to a sip
> > where the agent is logued in. Debuging we found that the invite
> > request to the sip ata sometimes takes from 20 seconds to 1 minute to
> > be sended.(checked with ethereal).The system never goes more than 30 %
> > of cpu usage, and when this happens, we noticed that the load average
> > is always over 1, even when the cpus are 99% idle and there is no
> > other process running in background, only asterisk.
> > Also, we checked that there is no packet loss and that the digium
> > card isn't producing spikes.
> >
> >
> > Anyone have ideas of what is happening?
> > Regards.
> > Ivan Cetta.
> >
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