[Asterisk-Users] Re: Asterisk Jitters

[Steven Critchfield]

On Thu, 2003-09-04 at 01:43, Zak wrote:
> I have three fxos from Digium installed in the box. 
> The Box got Pentium 4 2.4 Ghz and  512 RAM.
> I had the box working fine once but it stopped working (jitters) after
> a reboot.

Did you make sure the zap card drivers are loaded? Have you checked your
IRQs to make sure they didn't wander and start causing problems?

> ####
> If you have one, and the card is up and running, then it would be used
> for timing. Basically it is just needed in this case to make sure
> asterisk keeps chugging along at a known speed.
> 
> What speed hardware are you using?
> > Date: Wed, 03 Sep 2003 21:05:04 -0700
> > From: Zak <zakforever at netscape.net>
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Re: Asterisk Jitters
> > Reply-To: asterisk-users at lists.digium.com
> > 
> >   
> > Hi Steven,
> > 
> > I have a zap device installed in the box but I'm not sure if that's the one used for timing.
> > 
> > thanks.
> > 
> > Zak
> > 
> > 
> > 
> > Subject: Re: [Asterisk-Users] Asterisk Jitters
> > From: Steven Critchfield <critch at basesys.com>
> > To: asterisk-users at lists.digium.com
> > Date: Wed, 03 Sep 2003 11:17:15 -0500
> > Reply-To: asterisk-users at lists.digium.com
> > 
> > Do you have a zap device for timing?
> > 
> > On Wed, 2003-09-03 at 17:48, Zak wrote:
> > 
> >   
> > > > Hi,
> > > > 
> > > > Every time I dial into my asterisk box i hear nothing but asterisk 
> > > > jittering.
> > > > The following is an example of what I get on the asterisk CLI
> > > > 
> > > > Thanks
> > > > 
> > > > *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT 
> > > > on RTP
> > > > to 0
> > > > DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
> > > > DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user 
> > > > 'xirak' is 1
> > > >  out of 0
> > > > DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: 
> > > > Contact hop
> > > > : <sip:192.168.7.3>
> > > >     -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack
> > > > DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format 
> > > > changed from U
> > > > NKN to ULAW
> > > > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
> > > > timer at 16
> > > > 0 sample intervals
> > > >     -- Playing 'vm-login'
> > > > DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping 
> > > > retransmission on
> > > > '6E5D898E-492D-400B-A42B-8B25FE25F2EE at 192.168.7.3' of Response 1: Found
> > > > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
> > > > timer at 0
> > > > sample intervals
> > > > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
> > > > timer at 0
> > > > sample intervals
> > > > WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): 
> > > > Couldn't read u
> > > > sername
> > > >   == Spawn extension (extensions, 1001, 1) exited non-zero on 
> > > > 'SIP/xirak-259d'
> > > > DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak)
> > > >       
> >   
> >   
-- 
Steven Critchfield <critch at basesys.com>