Should be fixed...
-- Luke
From: Michiel Betel [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip show registry broken?
To: [EMAIL PROTECTED]
Date: Tue, 18 Mar 2003 10:17:56 +0100
Organization: Betel Consultancy
Just got the last CVS (Asterisk CVS-03/17/03-10:01:18) which works fine except
or perhaps use an official kernel from kernel.org?
On Friday 21 March 2003 04:37, Mark Spencer wrote:
sounds like you need to update your kernel-source RPM as well.
Mark
On Thu, 20 Mar 2003, Frank Hoonhout wrote:
I am in the process of trying out interesting software.
I setup Redhat
Carlos Crembil wrote:
Thank you Michael!.
I've applyied the configuration you sent me, but I have some troubles with
it, specially in the oh323.conf file. Lines like [register], [codecs]
are not appearing in my original file, and when I use this, asterisk
returns me an error and it fails to
I'm going through the same troubleshooting re: similar modprobe errors possibly
compile/kernel related
I'm using RH 7.3 and was trying to use alsa to detect my onboard sound card and was
following the instructions at
http://www-ccrma.stanford.edu/planetccrma/software/installkernelandsound.html
I have got my box up and running with a X100P and a S100U
but I found a bit of a funny..
I took the default config files and commented every line
and then I started creating my own config using the
commented out lines for reference.. (best way to learn)
None of my configs worked and I could not
How about a web interface module for asterisk itself, a webmin module
would be wonderful.
Regards
MIKE
On Fri, 2003-03-21 at 10:24, Mark Spencer wrote:
Dear Asterisk Community,
Due to the loss of our dear CVS and database server, the fact that the old
asterisk web site was pretty lame
i'd have to agree, but i'd suggest focusing on a context that was
probably needed that you commented out. by chance are you
running it as
./asterisk -vvvgc
and checking the error/warning messages on the console?
Steven Critchfield wrote:
On Fri, 2003-03-21 at 10:07, WipeOut . wrote:
I
On Fri, 2003-03-21 at 12:38, Brian Capouch wrote:
But the link in Mark's mail to the pdf of the rev II manual comes up
Cannot find link target or somesuch.
Is there something wrong with the server, or is it on my end?
B.
I was able to sucessfully download it this morning.
it seems as though this is the week of weeks for mark/digium.
mark just mentioned that while doing some work on the other
suites in the building, bell cut their T1 line. (it's back up
now)
but considering the week digium's had, maybe we should just 'go
with the flow' till next week, then start
In case I typed it wrong:
http://www.digium.com/handbook-draft.pdf
Mark
On Fri, 21 Mar 2003, Brian Capouch wrote:
But the link in Mark's mail to the pdf of the rev II manual comes up
Cannot find link target or somesuch.
Is there something wrong with the server, or is it on my end?
B.
Unfortunately, I'm kinda commited to NetMeeting. The problem is that I have
friends with a mix of Windows and Linux machines. Between Netmeeting and
Gnomemeeting, I should be able to get everyone connected.
So, I could try to find a Linux SIP client that looks and feels like MSN, or I
can
Do you know for sure whether the PBX issues a call termination pulse (ie
zero or reverse battery) on completion of a call?
Iain
--On Friday, March 21, 2003 8:56 pm +0100 Florian Overkamp
[EMAIL PROTECTED] wrote:
Hi guys,
So, now I've made a small demo box to do some IVR apps and hooked
Good idea!
Only 1 info : the site could be written in any form
(static html, php+mysql or so) or you prefer
to have it built onto a cms like postnuke o derivates?
matteo
Il ven, 2003-03-21 alle 16:24, Mark Spencer ha scritto:
Dear Asterisk Community,
Due to the loss of our dear CVS and
Mike Diehl wrote:
Unfortunately, I'm kinda commited to NetMeeting. The problem is that I have
friends with a mix of Windows and Linux machines. Between Netmeeting and
Gnomemeeting, I should be able to get everyone connected.
Good guy why? Why can't you use a decent H.323 client like
What are module versions on the kernel?
I did a google but didn't find an explanation that I understood.
With as much a I hate RH and what they do to kernels, I don't
think the kernel itself is the problem, nor will changing the
version solve your problems.
use depmod -ae so you can
On Fri, 2003-03-21 at 15:39, Brian Johnson wrote:
What are module versions on the kernel?
I did a google but didn't find an explanation that I understood.
From the kernel itself...
CONFIG_MODVERSIONS:
Can anybody explain the ${RDNIS} variable purpose and usage.
Many thanks
Adrian Brown
I've been looking at Asterisk now for several weeks and have had success using it with
SIP based soft phones on a local network. I'm having some issues going to the next
level (i.e. designing the system I want). Essentially this is what I want to do
PSTN -PRI- Asterisk Satellite --- VoIP
On Fri, 2003-03-21 at 17:02, Adrian Brown wrote:
Can anybody explain the ${RDNIS} variable purpose and usage.
Many thanks
DNIS: Dialed Number Information Service.
What number the caller dialed. Used when multiple numbers point to the
same line, or group of lines.
I'm pretty sure of what
Brian
Got it working some what, recompiled the kernel.
Now I get this:
[EMAIL PROTECTED] sbin]# ./asterisk -c
Asterisk CVS-03/20/03-16:56:24, Copyright (C) 1999-2001 Linux Support
Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
hi i sent gary an email about those fcc #'s. no response yet.
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On Friday 21 March 2003 1:49 pm, Jeremy McNamara wrote:
Mike Diehl wrote:
Unfortunately, I'm kinda commited to NetMeeting. The problem is that I
have friends with a mix of Windows and Linux machines. Between
Netmeeting and Gnomemeeting, I should be able to get everyone connected.
Good guy
GSM works but the voice quality is absolutely terrible. This is the
case with or without the prefix. (Did anyone ever figure out
whether is a toggle?)
One thing I didn't realise until reading the new documentation is that
the codec list is in order of preference. So, if there's an
You mean Greg? He will be on vacation this coming week but Call me (x
6275) and I'll try to find them for you.
Mark
On Fri, 21 Mar 2003, d hinton wrote:
hi i sent gary an email about those fcc #'s. no response yet.
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I'm having some problems getting an ATA186 behind NAT working. When I had
it on the same subnet as the Asterisk server, it worked fine. Now Ive
taken the ATA on the road with me, and it's behind a Dlink router+firewall,
doing NAT. I pick it up, hear a dialtone .. the firewall on the asterisk
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
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It is the responsibility of your device (SJpnone, mic/speaker pc) to
handle it's half of the echo problem (prevent what is playing in the
speaker from being picked up in the microphone.
I played with sjphone months ago with mic speakers and experienced the
same trouble. I would expect it to
have you tried nat=1 in your friend declaration? I notice in your dump it
says non-NAT
Mark
On Fri, 21 Mar 2003, denon wrote:
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
Thanks -- I didn't realize that needed to be set. It works now, but
there's a horrible echo on the sip client side. (I dont know about the
other side, as I havent called any humans yet :)
I don't, however, hear an echo when I call voicemail or such .. so I'm
assuming it's something with the
I am using a plantronics headset.. so as far as I know the echo is not
being caused by the mic picking up what is coming out of the headphones.
I have looked at all the settings in the software and I can't find any
echo cancellation features..
Oh well, guess its time to look for another
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