Hello,
Probably obvious to most of you but I’m still learning VOIP concept(and
telephony inside organization in general)
I have bunch of unrelated questions on my mind :)
- I picked SPA504G phones as “phone to go” for our small company. They are
cheap and available. Any pros/cons you can come
OK thanks a lot for your help now all is ok :)
2011/5/31 salaheddine elharit salah.elharit...@gmail.com
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta maheshka...@flexydial.com
Remove the _
Watch out the dialplan sequence
you have 1 then n, n , anf finally a 2. try changing the last 2 for n
if you are only dialing the number
0678922645
You have to remove the leading _ (underscore) and the ending point . ,
they are used only when dialing regular patterns. in your case wil
be
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta maheshka...@flexydial.com
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0678922645
Did you try different number in place of 5? I meant 1 2 etc..
Also check cli logs on console
Are you dialing from softphone or hardphone because some phone has
dialing regex for security.
--
Sent from my iPhone
On May 30, 2011, at 1:30 PM, salaheddine elharit salah.elharit...@gmail.com
On Mon, 30 May 2011, salaheddine elharit wrote:
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645,2,Hangup()
A better subject will get better responses.
Just a quick glance shows that you either mistyped your dial plan or you
need to read up on dial plan pattern
@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] please help
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten =
Hi All
Please help me in configuring asterisk for the below scenario:
I want to make calls to some mobile users then
My query is simple, my agent attends an incoming call from Queue and I
want my agent to play a recording and request the user to input 4 digit
input from his keypad and that
Hi All,
When Mobile user attends an incoming call from asterisk then I want to play
a recorded voice
and then I want to get four digit request by following voice instructions
from the user mobile keypad and that input can be stored.
I want to know technically whether this scenario is possible or
:35 PM
Subject: Re: [asterisk-users] Please help test the gender detection
moduleat575-613-4392
2009/2/18 Asterisk Asterisk nt_aster...@yahoo.com
Thanks for the feedback. I did some research and it looks like you were
calling over international lines. It also appears
On 22/02/2009 6:35 a.m., Asterisk Asterisk wrote:
If anyone needs the data output by this module (test data for the past few
days), I'd be happy to share. It includes the date/time, gender detected,
whether that's correct according to the tester, the winning ratio, and the
energy levels
On Mon, Feb 16, 2009 at 2:29 PM, Asterisk Asterisk
nt_aster...@yahoo.com wrote:
I need your help: please help test the gender detection module at
575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only takes
a second. I've been showing 90%+ accuracy and I want
to
On Fri, Feb 20, 2009 at 11:16:06AM -0800, Asterisk Asterisk wrote:
You have some good points.
Justin Newman isn't exactly someone we don't know. However I only
I agree that my name wasn't clear, but I was trying to avoid getting a
bunch of spam myself. I'm not sure if I've personally ever
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 7:27:12 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Hi,
Asterisk Asterisk wrote:
We've had a 65% success rate across
, February 21, 2009 1:17:48 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
On Fri, Feb 20, 2009 at 11:16:06AM -0800, Asterisk Asterisk wrote:
You have some good points.
Justin Newman isn't exactly someone we don't know. However I only
I agree that my
What's your caller ID? :)
Thanks for your help.
From: Darren Wiebe dar...@aleph-com.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 9:08:03 PM
Subject: Re: [asterisk-users] Please
On Sat, Feb 21, 2009 at 09:35:46AM -0800, Asterisk Asterisk wrote:
I'd also like to collect more information, including age and zip.
Figuring out how to do these things without affecting the tests
seems to take more time than the code itself.
ZIP would be meaningful to you if the caller is
)
justin_newman (yahoo im)
From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Sent: Saturday, February 21, 2009 11:42:41 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
On Sat, Feb 21, 2009
: Wednesday, February 18, 2009 4:01:15 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Hi Justin,
How far is your work from being able to do speaker verification? Not
*identification* mind you, but being able to tell that a captured voice is
the same
, February 18, 2009 4:13:46 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk Asterisk wrote:
Steve,
Tried
: Thursday, February 19, 2009 5:18:07 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I tested, even with the TV
loud in the background.
It got me wrong twice, but so do about 30
@lists.digium.com
Sent: Thursday, February 19, 2009 5:18:07 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I tested, even with the TV
loud in the background.
It got me wrong
On Thu, Feb 19, 2009 at 5:54 PM, Asterisk Asterisk
nt_aster...@yahoo.com wrote:
You sure you don't have a pony tail? :) Hehe.
It happens to the best of us. Hopefully after my fine tuning it will happen
to less of us!
I'm a male, your system told me I'm a male! Works!
The beep is the best!
Slightly off-topic,
On Mon, Feb 16, 2009 at 10:29:57AM -0800, Asterisk Asterisk wrote:
I need your help: please help test the gender detection module at
575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only
takes a second. I've been showing 90%+ accuracy and
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 19, 2009 5:18:07 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I
...@xorcom.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 10:48:10 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Slightly off-topic,
On Mon, Feb 16, 2009 at 10:29:57AM -0800, Asterisk Asterisk wrote:
I need your help
asterisk-users@lists.digium.com
*Sent:* Thursday, February 19, 2009 5:18:07 PM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I tested, even with the TV
loud
Asterisk Asterisk wrote:
You have some good points.
Justin Newman isn't exactly someone we don't know. However I only
I agree that my name wasn't clear, but I was trying to avoid getting a
bunch of spam myself. I'm not sure if I've personally ever spammed the
list and I'm pretty
2009/2/18 Asterisk Asterisk nt_aster...@yahoo.com
Thanks for the feedback. I did some research and it looks like you were
calling over international lines. It also appears that there was high than
average static on the line, which is not normal for my system. It's true
that I threw my
Darren Wiebe wrote:
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
I am a male and was detected as a male, so I'm feeling a bit left out. :-p
___
-- Bandwidth and Colocation Provided by
@lists.digium.com
Sent: Wednesday, February 18, 2009 4:13:46 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk
*From:* Steve Totaro stot...@totarotechnologies.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Sent:* Wednesday, February 18, 2009 10:57:47 AM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I tested, even with the TV
loud in the background.
It got me wrong twice, but so do about 30% of the people who call.
Ira
___
-- Bandwidth and Colocation Provided by
...@yahoo.com
Sent: Feb 18, 2009 4:09 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat
575-613-4392
Am Montag, den 16.02.2009, 11:45 -0800 schrieb Asterisk Asterisk:
Let me know how it works when you try the test number at 575-613-4392.
Hi Justin,
I tried your
:05:24 AM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
That's an interesting module.
Care to elaborate on what you designed it for ?
Thanks,
Ron
On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
I need your help: please help
@lists.digium.com
*Cc:* Asterisk Asterisk nt_aster...@yahoo.com
*Sent:* Monday, February 16, 2009 11:05:24 AM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
That's an interesting module.
Care to elaborate on what you designed it for ?
Thanks,
Ron
From: Steve Totaro stot...@totarotechnologies.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 18, 2009 10:57:47 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Totaro stot...@totarotechnologies.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 18, 2009 10:57:47 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
On Wed, Feb 18, 2009
LaCoursiere j...@jeff.net
Date: Thu, 19 Feb 2009 00:01:15
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Cc: nt_jnew...@yahoo.com
Subject: Re: [asterisk-users] Please help test the gender detection module
at 575-613-4392
Hi Justin,
How far is your work
Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Sent:* Wednesday, February 18, 2009 10:57:47 AM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro
stot
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 16, 2009 9:19:20 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Looks like my provider is not passing dtmf correctly .. Had a serious
laugh, system kept asking me if I
: [asterisk-users] Please help test the gender detection module at
575-613-4392
That's funny. The way I have it phrased, when I called I started talking to it
as well! I have some code for short list voice recognition and thought about
detecting yes and no in there, but I ran out of time
: Tuesday, February 17, 2009 12:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: nt_jnew...@yahoo.com
Subject: Re: [asterisk-users] Please help test the gender detection
moduleat 575-613-4392
That's funny. The way I have it phrased, when I called I started talking
-users] Please help test the gender detection moduleat
575-613-4392
After helping out it seems I’ve been determined a female(wrongly).
It was disappointing and I’m considering a visit to the Dr Phil Show to
work out my anger….
___
-- Bandwidth
I need your help: please help test the gender detection module at 575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only takes a
second. I've been showing 90%+ accuracy and I want
to make sure it's working correctly. Rain and significant background noise
seems to
nt_jnewman at yahoo.com
From: Ron Joffe ron.jo...@gmail.com
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk nt_aster...@yahoo.com
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613
--
*From:* Ron Joffe ron.jo...@gmail.com
*To:* asterisk-users@lists.digium.com
*Cc:* Asterisk Asterisk nt_aster...@yahoo.com
*Sent:* Monday, February 16, 2009 11:05:24 AM
*Subject:* Re: [asterisk-users] Please help test the gender detection
module at 575-613-4392
That's
: [asterisk-users] Please help me finding good A-Z provider
teliax? teliax.com
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied
by my current provider. Does not matter what protocol it uses. I'm
looking for good rates, stable quality
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied
by my current provider. Does not matter what protocol it uses. I'm
looking for good rates, stable quality and not so big prepayment required.
Thanks to all.
We do that.
www.direct-internet.co.in. BTW whats your location and where is your
major calling.
Lee Jenkins wrote:
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not
satisfied by my current provider. Does not matter what protocol it
uses. I'm
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied by my
current provider. Does not matter what protocol it uses. I'm looking for
good rates, stable quality and not so big prepayment required.
Thanks to all.
___
teliax? teliax.com
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied
by my current provider. Does not matter what protocol it uses. I'm
looking for good rates, stable quality and not so big prepayment
required.
Thanks to all.
No, I had very very bad experience with Teliax. I recommend Inphonex
(http://www.inphonex.com/).
Thanks,
Chandra.
Anthony Francis [EMAIL PROTECTED] wrote: teliax? teliax.com
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied
by my
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings depending on the
caller id. I have one set up and working for 'Internal'
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings
Hi,
sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50
active SIP channels so I
dont know if calls are getting dropped or not. Should I be worried?
2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided
deadlock for
'0xb7341470', 10 retries!
Hi to all,
I've a segmentation fault while using asterisk relatime conf with mysql db.
I've cretate sip_buddies and extensions tables into db and edit
res_mysql.conf, extconf.conf without any issues.
So when I start asterisk and my phone try to register using sip user
configured in my db,
On Tue, Oct 17, 2006 at 03:42:55PM +0200, flavio wrote:
Hi to all,
I've a segmentation fault while using asterisk relatime conf with mysql db.
I've cretate sip_buddies and extensions tables into db and edit
res_mysql.conf, extconf.conf without any issues.
So when I start asterisk and my
Hi I have 5 telular mod SX5e https://www.telular.com/v2/html/products/product_display.asp?productID=94 is a great stuff, but i have a extrange problem with asterisk. Sometimes the sound is ugly or choppy. The telular alone work fine all the time
For example if i made 5 calls from asterisk to gsm
I have a working Asterisk 1.2.5 system with SPA-3000 setup with the
SPA3000 Configuration Wizard for Asterisk from Voxilla.com.
I can make outbound calls from the Sipura POTS phone (not sure they are
actually going through the Asterisk box) but cannot get inbound calls
from the outside.
Hi Khairul
We are using 1.2.9.1 and polycom 1.6.6 firmware for
Polycom 601, the problem still exists. It only works for a while and then the
problem happens. But when we reboot the phone, it will work for another while.
I saw a thread about this problem in Digium bug report
page
Hi,I found your post onhttp://threebit.net/mail-archive/asterisk-users/msg04580.html
I am having the exact same issue with the Polycom IP601 (SIP version1.6.6.0036) with Asterisk
1.2.7.1.I was wondering if you found any solution to it. I would really appreciateif you could share your
Hello,
Thank you for the job well-done.
I installed the chan_h323 of the asterisk-1.2.7.1 and with lib
pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed
g729 from digium.
However, I am having a very funny behavour.
1. If I send a call on its ringing at the called
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to
Thank you for your quick response. I have successfully implemented
Intercom (Dialling within my office LAN) using Asterisk. To implement this,
I am using X-Lite Softphone.
Now, I want to make calls to US using VoIP Asterisk. I think that there is
no need of any external hardware to
hello,
Is there somebody wit a h323 terminal ?
___
Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos
services préférés : vérifiez vos nouveaux mails,
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Please help.. I need a h323 user for tests
hello,
Is there somebody wit a h323 terminal ?
___
Faites de Yahoo! votre page d'accueil
Hello all,
Iam using this Asterisk server since three weeks and i have to clarify some thing about Asterisk
here is my problem Iam trying to use my Asterisk as a gateway to pstn
and SER as a proxy and redirection server so,here in SER i had added
three or four users by using the command serctl
Hello,
IMHO, there are 2 ways to do this,
1) You can connect your VoIP modem to your asterisk box using x100p FXO card, you'll need to get one and install it properly.
2) Get SIP/IAX account from any VoIP provider and use it with asterisk.
Hope this helps.
Regards,
Umair Bari
On 5/12/06,
2006/5/12, Crazy Boy [EMAIL PROTECTED]:
I am unable to understand where to give above mentioned values? What
configuration files I should use to implement this using the Vebtel SIP
provider? Do I need to provide any more values to implement this using
Asterisk from Vebtel?
In addition to your
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to
I have a wired problem, i can recive call but i can't make any call.
ATT say that is my problema because the call is operator mode (???)
The log is
MFC/R2 Chan 1: Call control(1)
MFC/R2 Chan 1: Make call
MFC/R2 Chan 1: Making a new call with CRN 32772
MFC/R2 Chan 1: 0001
-
[1/
1/Idle
/Idle ]
[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Thu, 30 Mar 2006 17:13:46 -0400
Delivered: Thu, 30 Mar 2006 19:03:15
Subject:[Asterisk-Users] Please Help Test Quad PRI
Using NFAS
Please help me test my
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port T1 with NFAS setup.
I can dial-out but I cannot dial any 800 numbers (Global Crossing says I
need LDS service and that will be a couple weeks) so I cant test it
myself. I need at least
: Thu, 30 Mar 2006 19:03:15
Subject:[Asterisk-Users] Please Help Test Quad PRI Using NFAS
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port T1 with NFAS setup.
I can dial-out but I cannot dial any 800 numbers (Global Crossing says I
Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Thu, 30 Mar 2006 17:13:46 -0400
Delivered: Thu, 30 Mar 2006 19:03:15
Subject:[Asterisk-Users] Please Help Test Quad PRI Using NFAS
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port
: [Asterisk-Users] Please Help Test Quad PRI Using NFAS
hint... - listen to the queue for a bit
On 3/30/06, Melcon Moraes [EMAIL PROTECTED] wrote:
Are you gonna answer me? I'm the first in line and no answer! :)
[]'s
MM
-Original Message-
From: Steve Totaro [EMAIL
The music is cool, is it XM or a MP3 list?
Sounds fine, balanced if Alice wasn't talking all the time.
On 3/30/06, Steve Totaro [EMAIL PROTECTED] wrote:
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port T1 with NFAS setup.
I can
:
Sent: Thu, 30 Mar 2006 17:35:13 -0400
Delivered: Thu, 30 Mar 2006 16:31:58
Subject:[Asterisk-Users] Please Help Test Quad PRI Using NFAS
I apologize for the music on hold in advance ;-)
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Andrew Latham
Regards!
During the use of areski a2billing software
I'm getting same problem all the time.
Actually, after 15 minutes ofspeaking
to someone over calling card, connection brakes.
Installation was as smooth as it could be
so I don't think I made same kind of a mess in that domain.
Hi d_pejic,
First of all, please never send several times the same question to the
list, it's really
not respectful for the others. Your issues should not pass in priority
from others.
As Kpfleming pointed out, Add-Ons/A2Billing are off topic for this
list, so please redirect
add-ons question
hi all,
I am new to asterisk and want to configure it to work with current pbx.
Let me know how to delay rest of numbers from first digit.
What i want to do is,
Mynumber = 0714287895
my office pbx want us to wait few seconds before dialing outside number.
To access out side number we want
I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried is
*CLI -- Executing BackGround(SIP/1235-98f6,
/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm) in new stack
Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File
look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that
directory.
On 1/14/06, Abhishek [EMAIL PROTECTED] wrote:
I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried is
*CLI -- Executing BackGround(SIP/1235-98f6,
: Friday, January 13, 2006 6:35 AM
Subject: Re: [Asterisk-Users] Please help
look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that
directory.
On 1/14/06, Abhishek [EMAIL PROTECTED] wrote:
I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried
having these file in th
especified location with all th epermissions on it.
- Original Message -
From: Tom Vile [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 13, 2006 6:35 AM
Subject: Re: [Asterisk-Users
Thanks a lot. I was using the extension after the file name.
- Original Message -
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 13, 2006 7:12 AM
Subject: Re: [Asterisk-Users] Please
: [Asterisk-Users] Please help
look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that
directory.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
Hi,
I'm new in writing AGI script and actually newbie in Asterisk. I'm
writing a small script that will read the number inputed by the
caller of the extension 123. First he will dial number 123 then a
voice prompt will be played (welcome) then he should press number on
the softphone and the
Can someone tell me what problem I am having with Zaptel on
a Suse 10 distribution?
cc -I. -O4 -g -Wall
-DBUILDING_TONEZONE -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall
It looks like you do not have the kernel source code installed. Go to
'Yast' and 'Install Software'. Look for the package called
'kernel-source'. It will install the source for your kernel. Then run
the 'Update Software' to make sure the kernel and the kernel source are
the same version. Then
Just when I was ready to work on the deployment of several
hundred Asterisk SIP phones, I bumped into a serious
potential show stopper: an echo.
I went to Astricon and attended the Echo Session.
I also happen to know some (not a lot) about impedance
matching, etc.
My problem is mainly related
Hi!
If you have problems with MusicOnHold or run Meetme, please, gives this
patch a try as it might help you. If enough people test this, it will
potentially included in the upcoming 1.2 release.
Here's the info on Mantis:
http://bugs.digium.com/view.php?id=5374
Please, provide feedback on
Hi,
I'm trying to install FreeTDS. I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS, but still can't get it to work.
I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found.
I downloaded
I have 1.2.0 beta 1 running and it works fine. My x100p returns caller id
with no problems. When I test the CVSHEAD callerid fails with checksum and
len 0 errors.
I can run with the cvshead of zaptel and libpri with beta1 but only the
beta1 source works for caller id. Any source after beta1
You might have better luck posting this question on Asterisk-Dev (on
how to disable checksum etc).
Will do.
For the benefit of the archive, it has also been suggested that I should
have rxgain at 0.0 and relaxdtmf=yes in zapata.conf. Unfortunately,
neither of those fixed the problem for me.
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems happy
enough, returning:-
server clidtest # ./clidtest /dev/zap/1
Number: 041222, Name: MOBILE
(that number's fake.) However, I'm not getting
You might have better luck posting this question on Asterisk-Dev (on
how to disable checksum etc).
On 8/5/05, Jon Whitear [EMAIL PROTECTED] wrote:
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems
1 - 100 of 156 matches
Mail list logo
