Message: 3
Subject: Re: [Asterisk-Users] Re: Asterisk Jitters
From: Steven Critchfield <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Wed, 03 Sep 2003 23:13:58 -0500
Reply-To: [EMAIL PROTECTED]
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?
Steven,
The zap card drivers are loaded correctly and there don't seem to be
any IRQ problem.
Here is what I get from /proc/interrupts.
I'm not sure the sound card and eth0 using the same IRQ could be
causing the problem?
CPU0
0: 17728993 XT-PIC timer
1: 44037 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 9776462 XT-PIC eth0, Intel ICH2
8: 1 XT-PIC rtc
9: 103179821 XT-PIC wcfxo
10: 217202079 XT-PIC nvidia, wcfxo, wcfxo
12: 524504 XT-PIC PS/2 Mouse
14: 165140 XT-PIC ide0
15: 281208 XT-PIC ide1
NMI: 0
ERR: 0
Zak
####
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 <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk Jitters
Reply-To: [EMAIL PROTECTED]
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 <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Wed, 03 Sep 2003 11:17:15 -0500
Reply-To: [EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' 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)
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