don't know if Asterisk can do it, but ethereal can. Ethereal is an open
source protocol analyzer. Download it from www.etheral.com
On Fri, 2005-08-19 at 02:55, Bohuslav Coufal wrote:
> Hi all,
>
> is it possible to monitor RTP protocol (latency, errors, ...) by
> Asterisk or other software.
On Thu, 2005-08-18 at 21:08, [EMAIL PROTECTED] wrote:
> Hello All,
> I was recently fighting with an optimum online connection in NY.
>
> I finally got in touch with someone that confirmed they are throttling
> my upload connection.
I know they watch for people doing peer to peer file sharing and
ht to be "which one can I install and learn the
fastest" and not "which one will support the most clients, or have the
most uptime". Having said that, there is one caveat - I would stay away
from Fedora Core 3 or Debian unstable or whatever newest release of any
version. Also
Look for a book coming out from O'Reilly in a few months or so. It's
being offered on Amazon (pre-release). I can't remember the names of
the authors, but I'm pretty sure they are some of the Asterisk
developers.
Jeff
On Wed, 2005-06-29 at 19:37, Robert Goodyear wrote:
> On Jun 29, 2005, at 3:4
alitme requires CVS HEAD?
TIA,
Jeff Heath
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of programs
successfully).
My question is this: is it fairly common that the CVS HEAD version
won't compile? Before I start digging deeper and troubleshooting, I
want to make sure that it wasn't just something broken in the program
and it didn't compile because
be created automatically.
I'm running Asterisk version CVS-v1-0-02/17/05-17:34:40
TIA,
Jeff Heath
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anges to the password were reflected just as you surmised.
So that solves my question below.
Thanks!
Jeff Heath
On Wed, 2005-05-11 at 19:33, BJ Weschke wrote:
> Looking at app_voicemail.c with the copy I have here, it looks like
> vm_change_password is trying to dynamically
mail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 => 1234,Jeff
4035 => 1234,Pam
>
> On 5/10/05, Jeff Heath <[EMAIL PROTECTED]> wrote:
> > Where are user's
mail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 => 1234,Jeff
4035 => 1234,Pam
>
> On 5/10/05, Jeff Heath <[EMAIL PROTECTED]> wrote:
> > Where are user's
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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Is there a GUI for Asterisk that has similar capabilities to the Cisco
Personal Communications Assistant. I looked at the user interfaces on
the Wiki and they all seem to fall a little short. Is anyone on the
list using something they really like?
TIA,
Jeff Heath
On Tue, 2005-05-03 at 14:15, Christopher Jacob wrote:
> Josiah Bryan,
> Any useful results from your "number of installed systems survey"? If so,
> could you email them to me off-list?
actually, could you e-mail them on-list (might be more appropriate for
Asterisk-Biz though).
__
Does anyone have any direct experience with these?
What do they cost per port?
Do they support most of the features of the original phone (i.e. if I
have a Meridian phone, do all the buttons like conference, flash, hold,
etc. work the same) ?
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Aster
you might want to also post this to asterisk-biz.
On Tue, 2005-04-19 at 14:01, Denis Galvão - iSolve wrote:
> Hi all.
>
> Im participating of a project(a huge one) that will study Asterisk as its
> PABX base system.
>
> They ask me: "Who is using Asterisk as its base PABX!?"
>
> Now I ask you
I had the gain settings too far down in the config
> > file, they had no effect.
> >
> > Make sure you stop and restart * after changing any of these settings.
> > A simple reload won't suffice (I even unloaded and reloaded the kernel
> > modules, just to be sur
x27;t help you with (I've
got lots of telecom experience, but little Asterisk experience) is
changing the settings in Asterisk to cancel it. The good news, though,
is that this is a straight-forward echo cancellation problem, and once
you find someone who knows what the right settings are, you sh
d we'll put together a business plan to
make a lot of money (seriously, if you can figure this out e-mail me at
[EMAIL PROTECTED]).
Realistically, I doubt there is much you can do except try to get a
different VoIP carrier.
- Jeff Heath
>
>
On Tue, 2005-04-12 at 15:28, Noah Silverman wrote:
> Hi,
>
> I tried, and still get an echo.
> I don't think the problem is with the zap interface. It must be on the
> asterisk or phone side.
>
> -N
>
Echo requires 2 phenomena: 1) reflected energy 2) enough delay that it
is discernable. Th
On Fri, 2005-04-08 at 17:16, Mike Robinson wrote:
> Snacktime, it sounds like you are trying to build a business case for
> the migration from Norstar to *. If so, there is a solution that makes
> the business case a no-brainer. There is a gateway that enables you to
> re-use all your Norstar phon
FYI --
I changed the phone's DTMF mode from "in-audio" to "via RTP (RFC2833)"
and that fixed the problem.
Jeff
On Fri, 2005-04-08 at 13:15, Jeff Heath wrote:
> I think maybe it has something to do with tones from my phone not being
> recognized. I just call
2005-04-08 at 12:48, Jeff Heath wrote:
> I'm having trouble checking voicemail. When I make a call and the
> recipient doesn't answer, the call goes to voicemail, and it's being
> recorded (I checked the files in
> /var/spool/asterisk/voicemail/from-sip/4035/INBOX).
&g
I'm having trouble checking voicemail. When I make a call and the
recipient doesn't answer, the call goes to voicemail, and it's being
recorded (I checked the files in
/var/spool/asterisk/voicemail/from-sip/4035/INBOX).
My problem is that I can't get access to the recorded message. I dial
the ex
look at www.signate.com
Jeff Heath
On Sat, 2005-04-02 at 11:24, ruben cuevas rumin wrote:
> Hi all,
>
> I'm a Telecomunication Engeenering student. I have to develop a VoIP
> apliccation using SIP protocol. I have to develop the SIP Server, and
> the SIP clients.
>
>
tic echo.
Hopefully, someone who knows more about Asterisk's echo cancellation
capabities will post a follow up about whether or not there are settings
in Asterisk that might help.
Jeff Heath
On Thu, 2005-03-31 at 10:36, Philip Siegrist wrote:
> Hi All,
>
> On my * server I am getti
ale from 1 to 5 with 5 being the best. Really, the scale is
1 to 4.5 (but that's a long explanation). Here are some codec MOS
scores:
codec MOS delay (msec)
G.711 4.1 0.75
G.729 3.9210
the difference between 4.1 and 3.92 is significant and so is the 10 msec
delay.
Jef
On Sun, 2005-03-27 at 12:08, Andrew Latham wrote:
> I heard a great solution at Linux World Boston. A rather talented
> young man mentioned using a IPV6 VPN on the IPV4 internet. IPV6
> supports QOS by default. Just VPN straight back to the CO and have
> your POP there so you only need one firewall
.. echo cancellation requires a lot of processing cycles. In the
PSTN, echo cancelers are hardware devices that use DSPs with the FFT
algorithms in silicon. Do a 2048 tap echo canceler in software for 100
simultaneous call streams and you'll burn a lot of processor cycles.
Echo is a complex
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