Hi,
I have a more general question.
Our group over 5000 employees world wide wants to do
a survey for all employees asking them if they are happy with the job, salary,
environment etc
Can I use an * where people can call a certain phonenumber,
go through voice menues and entering
,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mardi 20 décembre 2005 20:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR and db
Serge Schumacher a écrit :
Hi,
I have a more
Hi,
Do you think * could play around 300 voicemenu messages simoultanously?
Regs,
Serge
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know if I have made any critical
mistakes/assumptions.
Shawn
P.S Contarra Envox I know, Asterisk
I am learning.
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Serge Schumacher
Sent: Tuesday, December 20, 2005
4:23 PM
To: Asterisk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mardi 20 décembre 2005 23:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR and db
Serge Schumacher a écrit :
Thnx for the fast reply
No unfortunatly this is not an option, we would never be able to reach them
all with 48 hours.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mercredi 21 décembre 2005 00:04
To: Asterisk Users Mailing List - Non-Commercial
What platform should you suggest to use asterisk?
I tried with SUSE now all the time but there are too many
problems with the updates.
On is the development platform on which * is developed ?
Regards,
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- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime
First off, stop using HTML email.
What is in your extconfig.conf? What shows when you type realtime mysql
status?
-Matthew
- Original Message -
From: Serge Schumacher [EMAIL PROTECTED]
To: Asterisk Users Mailing List
I downloaded latest * stable complile it successfully but
when compiling the asterisk-addons the res_config_mysql.so
is missing.
I followed the
instructions on wiki for Realtime.
What did you do wrong ?
Thanx,
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Subject: Re: [Asterisk-Users] Realtime
Serge Schumacher wrote:
I downloaded latest * stable complile it successfully but when compiling
the asterisk-addons the res_config_mysql.so is missing.
The stable version of Asterisk does not have Realtime support
Console:
*CLI dial [EMAIL PROTECTED]
No such extension '650' in context
'from-sip'
Extentions.conf
[from-sip]
switch = Realtime/@realtime_ext
extconfig
realtime_ext = mysql,asterisk,extensions_table
res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
What control is it ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: vendredi 7 janvier 2005 11:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sip protocol question ...
Hi,
I'm tryinig to
Hi,
I checked out asterisk and asterisk-addon into /usr/src, went to
asterisk-addons and did 'make' as usual, but I get
common.c:1:29: asterisk/logger.h: No such file or directory
common.c: In function `decode_header':
I have the zaptel and libpri in the same folder and thy compiled correctly.
: [Asterisk-Users] What's wrong with compile
On Fri, 2005-01-07 at 00:12 +0100, Serge Schumacher wrote:
common.c:1:29: asterisk/logger.h: No such file or directory
common.c: In function `decode_header':
You don't seem to have asterisk's header files installed in
/usr/include/asterisk. Install asterisk
-07 at 03:25 +0100, Serge Schumacher wrote:
Well I have in asterisk/include/asterisks a lot of .h files if it's what
you
mean ?
[snip correctly bottom posted reply...]
Iirc during compilation of asterisk-addons it will search in
/usr/include/asterisk for .h files. If you have them installed
I just installed the lastest cvs version of asterisks and
addons but after this the chan_capi doesnt compile anymore with ne new
header files?
Anyone an idea ?
Thnx.
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Hi,
Jan 6 01:43:09 WARNING[12209]: pbx.c:796
pbx_find_extension: No such switch 'Realtime'
What does this message mean ?
Something wrong with the switch statement in my
extensions.conf or maybe is the module net correctly installed ?
Thnx.
to fill in some gaps on the Wiki tonight, take a look in the
morning.
http://www.voip-info.org/wiki-Asterisk+manager+API
MATT---
-Original Message-
From: Serge Schumacher [mailto:[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 8:38 PM
To: Asterisk Users Mailing List - Non-Commercial
How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: lundi 3 janvier 2005 01:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
Can someone see what's wrong here
How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: lundi 3 janvier 2005 01:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
Serge Schumacher wrote
Hi,
Where can I find a complete * manager api guide, the one one wiki is missing
informations like the monitor function for example,
Thnx
Serge
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Can it be that the MeetMe application is not installed by
default even if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for
extension (from-sip, 550, 4)
Regards,
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Hi,
Can someone see what's wrong here please ?
I've installed the ztdummy driver to enable meetme, put his in my
extension.conf
exten = 550,1,Answer
exten = 550,2,Wait(1)
exten = 550,4,MeetMe(18|Md)
exten = 550,5,Hangup
this in my meetme.conf
[rooms]
;
; Usage is conf = confno[,pin]
;
conf =
Hi,
I do not understand the difference between SIP and IAX, is it only two
different protocols or something more special.
The problem I have is that I've created two users
Aix.conf
register = users1:passwd1
register = user2:passwd2
[user1]
type=user
context=default
secret=passwd1
Might be related to the musiconhold files using different encoding rates ?
Just an idea, also a newbie :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paid Up
Sent: vendredi 31 décembre 2004 14:01
To: asterisk-users@lists.digium.com
Subject:
: Serge Schumacher [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 31, 2004 8:00 AM
Subject: [Asterisk-Users] IAX users
Hi,
I do not understand the difference between SIP and IAX, is it only two
different
www.flexcom.lu
- Original Message -
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: 12/5/2004 5:13:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing - which program are you using?
I want to play around with post billing. List of all phone calls, ...
Which program
Hi,
What can it be when I can establish a connection between two Softphones but no
voice is transfered ?
thnx
Hugo,
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Hi,
My installation at home use two NAT translations before it reaches the linux
box where Asterisk is running on.
I use DSL with a Wireless router which fwd all packets to an Windows 2003 box
an this windows box it NATing the UDP and RTC packets to my linux box.
If I try to connect to it
modem and let it run
Asterisk, firewall and routing and NAT and your wireless.
You will need two network interfaces on the linux box
Outside users just will not be able to get in otherwise.
--- Serge Schumacher [EMAIL PROTECTED] wrote:
Hi,
My installation at home use two NAT
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