Hi,
Seeking recommendations for a good quality hosted predictive dialer service.
Low volume, single agent US dialing.
Thank you.
Best regards,
Sans
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New
Hello,
I notice when upgrading from 1.6.2 to 1.8 that in the menuselect
app_mysql is indicated as deprecated.
If one wants to use the MySQL-command in the dialplan, how to do so if
app_mysql is deprecated ??
Kind regards,
Jonas.
--
Use ODBC. Check the func_odbc.conf configuration file.
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
**
Hello,
I notice when upgrading from 1.6.2 to 1.8 that in the menuselect
app_mysql is indicated as deprecated.
If one wants to use the MySQL-command in the dialplan, how to do
Will ODBC become the default then ?
I see no ODBC-command to use in the dialplan.
Jonas.
On 05/05/2012 11:12 AM, Leandro Dardini wrote:
Use ODBC. Check the func_odbc.conf configuration file.
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be
Hello,
notice in the console output beneath that there is a resultid 6 but it
can not be cleared :
[May 5 11:46:27] -- Executing [s@sub:3] MYSQL(SIP/vart-0336,
Connect connid localhost dialplan host Asterisk) in new stack
[May 5 11:46:27] -- Executing [s@sub:4]
I ask this because I find the MySQL status information a bit alarming
(2946 connections) :
mysql status
--
mysql Ver 14.12 Distrib 5.0.95, for redhat-linux-gnu (x86_64) using
readline 5.1
Connection id:2922
Current database:
Current user:root@localhost
SSL:
Hi,
Because I had an incoming caller ID issue, I plan to upgrade Dahdi to the
latest version, hopefully this will solve the caller ID problem.
I'm running Elastix 2.2 with Dahdi tool 2.4.1.2
It it safe to upgrade to the latest version of dahdi? What is the best
way to do it? through yum?
Many
Jonas Kellens wrote:
Will ODBC become the default then ?
As far as I'm aware, yes.
You need to make sure that unix-odbc development libraries are installed.
Under Mandriva, Mageia:
urpmi unixodbc-devel
Ubuntu/Debian:
apt-get install unixodbc-dev
And here is a good link to get you
On Sat, 5 May 2012, ABBAS SHAKEEL wrote:
I wrote this article for making the things easy for newbies Please have
a look and let me know if you suggest some thing.
Please have a look and suggest changes
I have been looking online for a definitive how-to on using Asterisk as a
SIP trunk termination point . I am seeing conflicting messages and
methodologies for doing this.
I am not going to use a commercial vendor for this trunk, it will be used in
testing out various customer scenarios and am
Thank you very much Steve. Thanks for the in detail recommendations. I will
make modifications and let you know.
On Sat, May 5, 2012 at 9:22 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 5 May 2012, ABBAS SHAKEEL wrote:
I wrote this article for making the things easy for
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
I use software phone to test it... when I dialed 501, I cant hear anything
for about 10
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