hi Motiejus,
Can you give a command for converting it to normal voice , in audacity.
also i tired with more users still problem persists ,
can i try with gsm format , what you say?
regards
Dhaval
2010/5/18 Motiejus Jakštys desired@gmail.com
Hi,
The record is not double faster, it's 50%
On Wed, 19 May 2010, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question ? wouldn?t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel on its screen.
Information displayed by FOP is richer (you
Hi, anyone know?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 17 May 2010 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adding a context from the console
Hi, is it
I doubt codec will change anything, but you can try.
In audacity, Effect - Change Tempo...
2010/5/19 DHAVAL INDRODIYA dhaval.it01...@gmail.com:
hi Motiejus,
Can you give a command for converting it to normal voice , in audacity.
also i tired with more users still problem persists ,
can i
Cheers. That is exactly what I need (I wonder how come I didn't find out that
app) :-)
Thanks! Alex
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, May 18, 2010 11:24 PM
To:
Hello
as I understand, nobody not used this option?
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Vardan wrote:
Has any
Hello list,
I am trying to test a scenario wherein two clients configured on two diffrent
boxes try to communicate with each other by means of Asterisk. The softphone on
both the boxes is zoiper. One of the boxes is Unix, and has the server running
on it. The other is Windows.
When I make a
the simple way i can see it is the following;let's say you have did starts
with 1708
[from-did]exten = _1708XXX,1,Answerexten
= _1708XXX,n,Queue(SALES,,)exten = h,1,Hangup
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562
2010/5/19 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Wed, 19 May 2010, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question ? wouldn?t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator
Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com
Dumb question -- wouldn't it be easier to monitor a web interface
than a phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel on its screen.
Information displayed by FOP
On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
Hi, is it possible to add a context from the console using the dialplan
command?
Yes, just add an extension to it. The context will be created as needed.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC:
Many thanks.
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hello
I am new to Asterisk. I want to know is there any way to get DTMF input from
the user in the Dialplan.
Regards
Taimur Hasan
-THQ- !!!ONE
_
Hotmail: Powerful Free email with
Use read application
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 19, 2010, at 9:42 AM, taimur hasan wrote:
Hello
I am new to Asterisk. I want to know is there any way to get DTMF input from
the user in the Dialplan.
Regards
Taimur Hasan
-THQ-
Read is best for multiple digit DTMF input. For Single-Digit DTMF, you can
use WaitExten.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Wednesday, May 19, 2010 12:00 PM
To: Asterisk Users Mailing List
Hi!
as I understand, nobody not used this option?
Would you like everyone on this list to give you an answer? ;-
If you have a _real_ question about the U option of Dial(), then post it,
and someone here might or might not have an answer.
Philipp
--
Hi!
If Snom user could also testify, that would be very interesting to
know.
While you can do it, I would not recommend to have two sidecars with a
snom 370. Especially the boot-up can become very slow if there are lot of
SUBSCRIBEs to be issued, even if you tweak some of the subscription
Thanks a lot...
Regards
Taimur Hasan
-THQ- !!!ONE
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 May 2010 12:14:15 -0500
Subject: Re: [asterisk-users] DTMF Input from the User
Read is best for multiple digit DTMF
input. For Single-Digit DTMF,
OK I will try to explain my problems, sorry for my English.
So, I want to indicate in dial plan, that the dialed channel is connectted.
For example(extensions.conf):
exten = s,1,Dial(ToSomeOne)
exten = s,n,Hangup
In this example, you can know about that the channel is connected after
if
Greetings List,Trying to interconnect with a new provider.. the require
a compliance with RFC 3261 so knowing less than needed about RFC
documentations.. i would like to know if Asterisk is actually in compliance
with RFC 3261 or not.. Can any one help with this?
Regards
--
Tarek Sawah
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come across is that the peer is congested and
sending RTCP messages asking us to slow the RTP down. Is there any way
we can verify this?
[May 17 13:42:45]
Sip debug peer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, May 19, 2010 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cause
What should I expect see if it is the peer asking us to slow down RTP?
Thanks again.
On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas da...@debsinc.com wrote:
Sip debug peer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
That is the RFC number for SIP. Yes, Asterisk is compliant with RFC. I am
not sure to what degree but I haven't ever faced non-compliance on SIP RFC 3261
ever with any provider.
-Bruce
On Wed, May 19, 2010 at 2:28 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings List,Trying to
Hello group,
I have asterisk running on my ubuntu machine, and I have a
peer to peer network with an XP machine, both of the running x-lite client, I
try
calling either of the soft phone from the other and the response I get is on my
asterisk console is as below:
[May 19 19:31:18]
On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime achera...@gmail.com wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to
The two phones belong to context phones and the two extensions are in context
internal. In context phones you need to include = internal so that context
phones knows about those extensions. Or put the two extensions in context
phones and not context internal.
--
Jim Dickenson
Thank you David. I was thing about the cisco solution but cost is the
issue as I will so many DSP to for this amount of calls.
Regards
Adolphe Cher-aime
From my Iphone
On May 19, 2010, at 4:23 PM, David Backeberg dbackeb...@gmail.com
wrote:
On Wed, May 19, 2010 at 12:38 AM, Adolphe
Adolphe-
Thank you David. I was thing about the cisco solution
but cost is the issue as I will so many DSP to for this
amount of calls.
If you're not doing G729 or other LBR codec (or encryption, or echo can with
long tail length, or other high level
requirement for RTP processing) then you
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to
Jonathan-
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com
wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
On Tuesday 11 May 2010 01:25:30 pm Tim Nelson wrote:
I have a handful of Asterisk 1.4.x installations where users dial 'outbound
calls' to the PSTN even though the destination is on the same Asterisk box
or on another Asterisk box on the same network. Instead of paying twice
for the call to go
The High Availability HASTerisk (HAAST) product on www.generationd.com is a
software solution that does automatic failover, etc between multiple asterisk
machines. I'm guessing this could be part of an overall solution for you
From:
Thank you Jonathan.
I really appreciate
Adolphe Cher-aime
From my Iphone
On May 19, 2010, at 6:26 PM, Jonathan Thurman
jonat...@thurmantech.com wrote:
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com
wrote:
Hello Everyone,
I must deploy
Jonathan for redundancy which software do you recomand?
Adolphe Cher-aime
From my Iphone
On May 19, 2010, at 6:26 PM, Jonathan Thurman
jonat...@thurmantech.com wrote:
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com
wrote:
Hello Everyone,
Thanks Michelle
Adolphe Cher-aime
From my Iphone
On May 19, 2010, at 8:02 PM, Michelle Dupuis mdup...@ocg.ca wrote:
The High Availability HASTerisk (HAAST) product on
www.generationd.com is a software solution that does automatic
failover, etc between multiple asterisk machines. I'm
On Wed, May 19, 2010 at 6:13 PM, Adolphe Cher-aime achera...@gmail.com wrote:
Jonathan for redundancy which software do you recomand?
Without knowing exactly what you are trying to do beside having at
least 500 outbound calls, that would be impossible to say. I mostly
use a home grown HA Linux
On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote:
On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote:
hi all,
I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue
Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record
audio clip with mixmonitor() as
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