I've been asked to forward this announcement to the list. It's a little
"short notice" as the meeting is this Wednesday night. I'm one of the
presenters as well :-)
From: Gary Poster <[EMAIL PROTECTED]>
Date: October 10, 2005 11:51:10 AM EDT
To: zope-announce@zope.org, python-announce-list@pyth
On Monday October 10 2005 08:18, Kanishka Somaratne spake:
> hi
> how much bandwidth is used for the following codecs
http://www.voip-info.org/wiki/index.php?page=Bandwidth+consumption
--
Joey Kelly
< Minister of the Gospel | Linux Consultant >
http://joeykelly.net
"I may have invented it, but
Hi Everyone,
Got a setup as follows:
Telco > Siemens HiCom 300E <> Asterisk1
Asterisk2 <> Siemens HiPath 4xxx
The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
Hello, i`m carlos, i`m just begining to use Asterisk at Home, so i have learned to configure a several extensions, but now i have a FXO target and i wanna to connect to PSTN, but i dunno how to do. i`d like to receive support from you...thanks
___
--Band
Rich Adamson wrote:
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt latency when doing a 'lspci -v'.. should that
number be 0? If so, does anyone know how to set that at boot time?
I played around a fair amount with the latency
There isn't a way to do it in agents.conf.
That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to s
I'd also like to through the into the discussion my recommendation for
www.xoops.org much better solution than Mambo as far shorter learning
curve.
Also larger development team behind Xoops, particularly as Mambo is now
split messily into two camps and the whole issue about who owns what.
For a d
>
> >
> > One other item to check is to ensure the digium T1 card is on its own
> > dedicated interrupt. Use 'cat /proc/interrupts' from the system command
> > line.
> >
>
> It is on one interrupt, first thing I checked when the problem cropped up.
> One
> thing I did notice was interrupt la
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am using this and it
appears to be better
Mike M wrote:
On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
Dinesh Nair <[EMAIL PROTECTED]> wrote:
too much divergence and we have two pieces of software competing for each
other.
My guess is that if they succeed, they will diverge significantly.
We wil
Ok :)
--
From: Rich Adamson[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, October 10, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] sip register incoming
... which is equivalent to my protocol=0x22 ;)
Nevertheless. I think it was a problem with chan_capi being too old
and not supporting protocol=0x22 layermask=0xf (it would not work
without layermask=0xf).
I am currently trying to get it working with chan_misdn. Will let you
know how it go
Rich Adamson wrote:
One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt latency wh
On Sunday 09 October 2005 18:42, Rod Bacon wrote:
> Can someone who is knowledgable in the traditional telco space please give
> me a layman's explanation (or point me to an appropriate url) of LBO as per
> the zaptel configuration file?
Unless something has changed in the last two years, zaptel t
I failed to make TDM400 working too myself. But I believed
I passed the the driver stuff... by installing zaptel-1.2.0-beta1.
Inside the package, there is a script "zaptel.init" that should
take care of loading/unloading the driver.
Min
> -Original Message-
> From: [EMAIL PROTECTED]
> [
On Mon, Oct 10, 2005 at 01:28:17PM +0200, lenz wrote:
>
> Hello list,
> I am looking for a way to have multiple remote Windows users download a
> package and get connected to *. My idea would be that they run a simple
> app, it connects without any setting to an * box (maybe via IAX) and then
On Mon, Oct 10, 2005 at 12:28:43PM +0200, [EMAIL PROTECTED] wrote:
>
> Hi Tzafrir !
>
> Thanks for your help!! Now it works.
Now, how would we detect that to avoid needless manual editing of the
CPU?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
Hi list (again),
I have another question which I have not been able to resolve from
neither the wiki nor Google.
I've been able to configure a multi-tenant setup of asterisk for 2
small call centers with no problem, by simply playing with contexts
(which I guess is how everyone else is do
Hi all!
I'm experimenting a strange problem in my Asterisk PBX:
I've got an Asterisk pbx in the office: I dial an external number; the dialled
number answers me correctly, but as soon as I press the '*' button (i.e. to
navigate through the menus or to enter a password) my Asterisk box put me on
Hi,
Yes, you can use the Fritz! in PTP mode, though only if you are using the
mISDN drivers. The mISDN driver should be called like this:
modprobe avmfritz protocol=34
Craig
- Original Message -
From: "Lionel Riem" <[EMAIL PROTECTED]>
To:
Sent: Monday, October 10, 2005 4:04 PM
Hi list,
I have a couple of questions related to asterisk billing and the
generation of cdr logs. I've searched the wiki but have not found my
answers, hopefully you guys can help.
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both?
Thank you for your respond, please see more detail inline...
Min
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Tzafrir Cohen
> Sent: Friday, October 07, 2005 4:57 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] TDM02B
> Sorry this is a bit of a newbie question, I've been at this for a few
> months and still have not quite figured this one out.
>
>
> I've been able to setup one itsp (incoming calls) (sip account) with a
> register line like this:
>
> register => nnn:[EMAIL PROTECTED]
>
> -or-
>
> regist
> > >Two asterisk boxes 150 miles apart, both cvs-head as of this morning
> > >(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
> > >C7960 calls exten on remote system (also C7960), and call goes to VM.
> > >No other calls in either system (eg, no load).
> > >
> > >Both bo
On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
> Dinesh Nair <[EMAIL PROTECTED]> wrote:
>
> >too much divergence and we have two pieces of software competing for each
> >other.
>
> My guess is that if they succeed, they will diverge significantly.
We will have two pieces of soft
On Mon, 2005-10-10 at 13:28 +0200, lenz wrote:
> I am looking for a way to have multiple remote Windows users download a
> package and get connected to *. My idea would be that they run a simple
> app, it connects without any setting to an * box (maybe via IAX) and then
> people press a butto
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register => nnn:[EMAIL PROTECTED]
-or-
register => nnn:[EMAIL PROT
> >If you don't have any T1/E1 connections to the outside world, then
> >pick one channel bank and call it your official source of sync, and
> >change the above definitions to sync off that channel bank. On all
> >other channel banks, configure them to sync off the asterisk card.
> >
>
>
>
> >Y
i have tellular cell phone plugged to fxo modules ,the
problem that i am facing is when i dial a number on
the fxo modules the call is been answered before it
picked up on the other side , i thought that the
analoge lines do not support the answer switching
feature but not tellular cell phones beca
On 10/10/2005, "Rich Adamson " <[EMAIL PROTECTED]> wrote:
>If you don't have any T1/E1 connections to the outside world, then
>pick one channel bank and call it your official source of sync, and
>change the above definitions to sync off that channel bank. On all
>other channel banks, configure t
>What am I doing wrong here? Why is this happening?
>
>libpri is version 1.0.7-1 (debian package) asterisk is version
>1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2
>
>
>-- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack
>-- Called g1/0916000739
>-- Channel
[EMAIL PROTECTED] wrote:
WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK8PChannel7IsClassEPKc
Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed!
Does anybody know which is the problem ?
It seems Asterisk source and binary version do not fit.
hi
how much bandwidth is used for the following codecs
723 r 5.3
723 r 6.3
723 r 8
what i know so far is the
723 r 5.3 uses 5.3 k up and 5.3k down
723 r 6.3 uses 6.3 k up and 6.3k down
729 r 8 uses 8 k up and 8k down
is this correct or is it like the following
723 r 5.3 uses 11 k up and
Andy,
I may be wrong, but I think that you need to edit the code and
recompile to change the message. I wanted to add a numeric line as
the first line of our system's voicemail pager message so it would
work with numeric pagers as well as text pagers. AFAIK, editing the
code and recompili
I am still looking to solve this problem, does anyone have any ideas?
Thanks,
Andy
-Original Message-
From: Andy Goss
Sent: Friday, October 07, 2005 5:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] call to a particular 800 number never
I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is
possible to customize the email message sent to the pager email address.
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
___
The UTStarcom F1000 with the latest firmware (3.10st) has improved
sound volume over the default firmware shipped with the units.
Also, TFTP configuration works well so you don't have to configure the
units with the keypad. You will need to get the configuration
compiler from your vendor and be a
I am trying to setup an h32h channel in Asterisk
Firstly I tried to use chan_h323, but I was not able to compile the
required pwlib ad open323 version under my system (Suse Linux 9.2)
Next I tried to use oh323. I succed in compile and install the
pwlib-Mimas_patch2-src-tar.gz
then
openh323-Mima
What am I doing wrong here? Why is this happening?
libpri is version 1.0.7-1 (debian package)
asterisk is version 1.0.7.dfsg.1-2 (debian package)
zaptel is version 1.0.9.2
-- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack
-- Called g1/0916000739
-- Channel 0/1, span 1
Four years ago, I faced a real dilemma in my business: the Visual Voice PRI
dll had a bug that considered unanswered any call after ringing for 20
seconds. This bug was in fact killing my business, because for international
calling, the setup of the call was already close to 20 seconds on many
case
What you are looking for is a pc2phone dialer. This can be preconfigured
with all settings and when it connects to your * it ask for username and
password or just a pin. There are many of these out on the net. Most is
however locked to a provider but you will also find many that you can buy
with yo
> I am a newbie to * and I am having a problem which appears strange as I did
> not find any mention of it anywhere in my search.
>
> Simply speaking, I have an external SIP proxy server which I am trying to
> configure for incoming and outgoing calls from my asterisk installation. So
> here is m
Hi all,
What methods (software or even on paper) would you folks use /
recommend for the purposes of documenting how a dial plan is
constructed? ie. what extensions jump to other extensions, etc? This
is as a means of getting the "big picture" rather than having pages
and pages of printed extensio
I don't think you will have any problems at all. I have always used the
lowest setting no matter how long the cable (never had a very long cable
run) and have never run into any problems.
My assumption of the CSU settings are because different CSUs have different
voltage outputs (this is just a
Hello list,
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects without any setting to an * box (maybe via IAX) and then
people press a button to talk. It would be okay if they had t
Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configu
> I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS),
> no PRIs.
>
> Some users are complaining that they hear clicks and pops on the FXS lines,
> generally when they pick up the phone it's noisy. This happens only after a
> while, e.g. after a fresh restart of everythi
I took the certification in Astricon Madrid, still I have to get any kind of
proof/certificate.
I contacted the testing company and they told me it was just a matter
of time, so probably they are working on this probably those are
just "super rapid growing" problems.
Regards!
s> The origi
Hi Tzafrir !
Thanks for your help!! Now it works.
It took some time to find everything and to set up everything, but now it
works.
So I can tell: using asterisk on book pc's with cyrix processors and VIA
chipset compiles fine.
Now I need to check what the performance is like.
thanks again,
Hello,
Well, now, with the help of mISDN you can, according to http://
www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-
html/x3343.html :
"With the introduction of the isdn4linux new mISDN architecture and
it's capi layer, that problem is fixed. chan_capi supports PTP
I think you can't use a Fritz Card for PTP. You need an active card. We use the
the beronet ISDN Cards with misdn.
Lionel Riem wrote:
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for quite
a while in PTM mode and it was working finely.
Now, I needed more DI
Hi, all
I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.
phonebox2*CLI> zap show status
No Zaptel interface found.
I assume that driver is not loaded, but I am sure I have installed it (I
compiled zaptel and then re-build asterisk and did make install
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for
quite a while in PTM mode and it was working finely.
Now, I needed more DID/MSN, so I switched to PTP. But now nothing
works anymore :(
I am using Asterisk on Debian Sarge stable and installed Asterisk
alo
The original poster's statement about not even receiving any proof that
he was certified is kind of amazing. That's not a certification
by any definition I know of. I would push Digium on that
because they really don't have a leg to stand on if that is true.
If they sold it as a certification t
asterisk-users
where can be find zaptel cvs change log ?
thanks
oncemore
[EMAIL PROTECTED]
2005-10-09
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
h
On Mon, Oct 10, 2005 at 07:32:31AM +0200, Tzafrir Cohen wrote:
> On Sun, Oct 09, 2005 at 07:37:41PM -0400, Steve Gladden wrote:
>
> > The debian package installs something else called "mpg321" and creates
> > an alias or symlink called mpg123 to mpg321.
>
> Get the package mpg123 from non-free
T
On Friday 07 October 2005 17:48, Michael Stahl wrote:
> Roman:
>
> I created two bash scripts called Mail2Fax and Fax2Mail for use with the
> asterisk sever.
>
> They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
> They make using these apps a lot easier, including being able to
101 - 157 of 157 matches
Mail list logo
