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 SoftPhone
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 br
I have a linejack and a phone jack in my asterisk server working well
between the SIP phones and the phonejack. what I cannot get to work is
the outbound linejack Phone/phone0 trunk line? how can I get a SIP or
Phone/phone1 phonejack phone to dial 9 then outside number and pickup
Phone/phone0 and
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
>
>
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 w
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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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
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.
>
> ___
> Asterisk-Users mailing l
> RDNIS: Reverse Dialed Number Information Service
> Essentially Caller ID, maybe not as susceptable to being changed though
> as I think it is transmitted in PRI signalling not as FSK on the line.
> Essentially to modify it, the caller would have to be able to convince
> the telco to send somethin
>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 a
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.
> G
hi i sent gary an email about those fcc #'s. no response yet.
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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]>
=
On Fri, 2003-03-21 at 17:26, Lenny Post wrote:
> 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 wan
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 wh
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 ---> V
Can anybody explain the ${RDNIS} variable purpose and usage.
Many thanks
Adrian Brown
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:
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
did this work for you?
Frank Hoonhout ([EMAIL PROTECTED]) wrote:
>
>After some more investigation, I think I know what I need to do.
>
>Taken from the kernel-how-to docs.
>
>3.4 The 'depmod' gives "Unresolved symbol error messages"
>When you run depmod it gives "Unresolved symbols". A sample erro
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 OpenP
I only have one context at this point, I am still getting
my head around the basics before I venture into multiple
contexts and macros and all the more advanced stuff..
I am running it ./asterisk -vvc but did not see any error
that pointed to the problem..
I guess the main thing is that its worki
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 da
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
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 set
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?
>
>
Hi guys,
So, now I've made a small demo box to do some IVR apps and hooked it up to
an analog line of an Ericsson MD110 pbx. Everything seems to work fine, but:
issue: even though X101P is configured for kewlstart it fails to see
disconnects unless I enable busydetect
issue: i don't get calleri
Havin changed machine (now running Debian 3.0r1) I newly have problems
compiling ztdummy.
I've again the *unresolved symbols* for ztdummy.
So, I've downloaded a now fresh 2.4.20 kernel source, compiled it
successfully enabling USB, ISDN, PPP, as modules, according to what I've
read in this ML and
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 w
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.
__
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:
> >
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 an
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.
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On Fri, 2003-03-21 at 10:07, WipeOut . wrote:
> 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.. (be
After some more investigation, I think I know what I need to do.
Taken from the kernel-how-to docs.
3.4 The 'depmod' gives "Unresolved symbol error messages"
When you run depmod it gives "Unresolved symbols". A sample error message is
given here to demonstrate the case:
--
On Fri, 2003-03-21 at 10:28, Frank Hoonhout wrote:
> Steven:
>
> Here is the output from depmod -ae
> I do see errors.
>
> [EMAIL PROTECTED] /sbin/depmod -ae
> depmod: *** Unresolved symbols in /lib/modules/2.4.18-27.8.0/misc/wcusb.o
> depmod: usb_control_msg_Re80c4f9a
> depmod: u
This is the last question for today I promise.. :)
I have both SJphone and eStara installed on my PC..
Both have an horrendous echo when making a SIP call
out of the X100P or to the phone attached to the
S100U..
Are there any parameters I can use in the sip.conf
or anywhere else to reduce the ec
Steven:
Here is the output from depmod -ae
I do see errors.
[EMAIL PROTECTED] /sbin/depmod -ae
depmod: *** Unresolved symbols in /lib/modules/2.4.18-27.8.0/misc/wcusb.o
depmod: usb_control_msg_Re80c4f9a
depmod: usb_submit_urb_R77d0e891
depmod: usb_set_configuration_R38118f
Does the 'T' option for the Dial application work?
I have the 't' option working fine but I want to be
able to transfer any call so need to specify 'Tt',
but the 'T' option doesn't seem to work..
Here is the line from extension.conf
exten => 1234,1,Dial(Zap/2,,Tt)
Thanks
--
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 no
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 anyway, and that collectively, we have
the web skills of a C programmer, we decided maybe it would be a good idea
to invite the Asterisk community to submit themes
On Fri, 2003-03-21 at 08:17, Brian Johnson wrote:
> I'm going through the same troubleshooting re: similar modprobe errors possibly
> compile/kernel related
> I'll try an older, stock RH kernel and let you know how it goes
With as much a I hate RH and what they do to kernels, I don't think the
ke
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
tha
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 st
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
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 f
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