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I have three fxos from Digium installed in the box. <br>
The Box got Pentium 4 2.4 Ghz and 512 RAM.<br>
I had the box working fine once but it stopped working (jitters) after
a reboot.<br>
<br>
<br>
####<br>
<pre wrap="">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?</pre>
<br>
<blockquote type="cite" cite="mid20030903214101.27843.98296.Mailman@digium.com">
<pre wrap="">Date: Wed, 03 Sep 2003 21:05:04 -0700
From: Zak <a class="moz-txt-link-rfc2396E" href="mailto:zakforever@netscape.net"><zakforever@netscape.net></a>
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: [Asterisk-Users] Re: Asterisk Jitters
Reply-To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
</pre>
</blockquote>
<blockquote type="cite" cite="mid20030903214101.27843.98296.Mailman@digium.com">
<pre wrap="">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 <a class="moz-txt-link-rfc2396E" href="mailto:critch@basesys.com"><critch@basesys.com></a>
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Date: Wed, 03 Sep 2003 11:17:15 -0500
Reply-To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Do you have a zap device for timing?
On Wed, 2003-09-03 at 17:48, Zak wrote:
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">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
'<a class="moz-txt-link-abbreviated" href="mailto:6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3">6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3</a>' 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)
</pre>
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<pre wrap=""><!---->
</pre>
<pre wrap="">
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