- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere
Hello,
We are going to start development for a product based over Asterisk.
According to you, which is the preferred language for AGIs / IVRs
development in Asterisk. I got opinions that Perl is going to be replaced by
PHP for all future developments.
--
Kashif Naeem
Business Development
Hello,
Is it possible for a system like Asterisk to create AOC messages or should
such AOC messages always originate in PSTN ?
Regards
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Drop Asterisk, move to Freeswitch. Much easier to interact with external
code bases, and it has more than one language interpreter built in
(javascript, lua, etc.).
If you're intent on staying on Asterisk, I would suggest skipping AGI,
and write a client that monitors the state of asterisk via
On Tue, May 5, 2009 at 1:34 PM, Kenneth Shaw k...@expitrans.com wrote:
Drop Asterisk, move to Freeswitch. Much easier to interact with external
code bases, and it has more than one language interpreter built in
(javascript, lua, etc.).
agreed but FS is newer and under test environment.
If
On Mon, May 04, 2009 at 07:55:04PM -0500, Atlanticnynex wrote:
I don't like DAHDI anyway... even if it is just the name. Gets me
confused with DUNDi and other fail acronyms.
What's there not to like about DAHDI? It's a fun game:
Kenneth Shaw schrieb:
Drop Asterisk, move to Freeswitch. Much easier to interact with external
code bases, and it has more than one language interpreter built in
(javascript, lua, etc.).
If you're intent on staying on Asterisk, I would suggest skipping AGI,
and write a client that monitors
Tzafrir Cohen wrote:
On Mon, May 04, 2009 at 07:55:04PM -0500, Atlanticnynex wrote:
I don't like DAHDI anyway... even if it is just the name. Gets me
confused with DUNDi and other fail acronyms.
What's there not to like about DAHDI? It's a fun game:
It doesn't support my linecard (I know
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
This is my first message to the list/newsgroup.
This weekend and after to have fought by some time with my soundcard
with respecto to the voice capture, after assuring to have solved that
problem, I installed Asterisk on Debian GNU/Linux
On Tue, May 05, 2009 at 10:12:54AM +0100, Thomas Kenyon wrote:
Tzafrir Cohen wrote:
What's there not to like about DAHDI?
It doesn't support my linecard (I know it's not digium's fault) so I'm
stuck using an old version of asterisk.
Which card is it?
--
Tzafrir Cohen
On Tue, May 5, 2009 at 9:52 AM, Kashif Naeem kas...@haditelecom.com wrote:
Hello,
We are going to start development for a product based over Asterisk.
According to you, which is the preferred language for AGIs / IVRs
development in Asterisk. I got opinions that Perl is going to be replaced by
On Tue, 5 May 2009, Kashif Naeem wrote:
Hello,
We are going to start development for a product based over Asterisk.
According to you, which is the preferred language for AGIs / IVRs
development in Asterisk. I got opinions that Perl is going to be replaced by
PHP for all future
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service ,
I was looking for a (http socket module / mysql module) not using
AGI(perl/php/shell) for asterisk in order to do intensive database / web server
interactions as needed without performance to much overhead.
Is there a real benefit in using : Asterisk cmd MYSQL app_addon_sql_mysql , I
see it
Hello list,
I wondered if those of you using Polycom phones could recommend a decent
cord detangler. I've had quite a few handsets get the tabs broken off in
the jack from cord detanglers due to the recessed nature of the jack.
This seems like it would work but I wanted some opinions before I
On 5 May 2009, at 15:02, Khaled W. Chehab wrote:
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big
problem
if the called party line is closed or number is incorrect or have a
voice mail (Early media 183) user will not hear the
On Monday 04 May 2009 20:12:19 Atlanticnynex wrote:
I'm trying to get asterisk cdr_odbc configured, but it can't connect
through my odbc driver.
switchboard*CLI module load cdr_odbc
[May 4 20:06:04] ERROR[17758]: cdr_odbc.c:358 odbc_load_module:
cdr_odbc: Unable to connect to datasource:
See if that helps you: http://edmundlong.com/edsBlog/odbc-cdr-with-asterisk/
Regards
On Tue, May 5, 2009 at 11:07 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 04 May 2009 20:12:19 Atlanticnynex wrote:
I'm trying to get asterisk cdr_odbc configured, but it can't
There is no perl instance used by MYSQL. It uses the mysql client library
Julien Chavanton wrote:
I was looking for a (http socket module / mysql module) not using
AGI(perl/php/shell) for asterisk in order to do intensive database / web
server interactions as needed without performance to
Hello list,
I recently started testing the chan_mobile addon and after a successful
installation and configuration I have a couple of problems that I can't fix
without your help.
I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm
packages) and a Nokia N80 phone.
hi Team,
I am stucking in this issue now. Can anybody do me a favor? I am trying to
get background noise related code. Do anyone has experenice which code is
responsed for generat/maintent background noise?
Regards,
Sunny
On Mon, May 4, 2009 at 11:14 AM, Sunny Du sunny...@gmail.com wrote:
The
Is it possible to make a call from a SIP/IAX softphone, say Zoiper, on
one computer to an Asterisk system without having an extension/account?
If so what are the terms I need to search for to figure out how to do
it? So far anonymous SIP has got me the closest I think but no brass
ring.
Hello Daniel,
You will find the information at http://www.voip-info.org/ and
http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the
Online Book link) very useful.
The asterisk package by itself should be adequate for SIP/IAX calls.
I don't think you need libpri unless you are
On Tue, 5 May 2009, Kashif Naeem wrote:
We are going to start development for a product based over Asterisk.
According to you, which is the preferred language for AGIs / IVRs
development in Asterisk. I got opinions that Perl is going to be
replaced by PHP for all future developments.
My
Hi there Daniel,
I havnt caught the beginning of this thread - but as we speak I'm in the
process of installing an Ubuntu server into a VM and getting 1.4 up and
running in a hope to replace my home system.
I did kinda follow a link
Un-top-posting...
On Tue, 2009-05-05 at 11:52 +0500, Kashif Naeem wrote:
We are going to start development for a product based over Asterisk.
According to you, which is the preferred language for AGIs / IVRs
development in Asterisk. I got opinions that Perl is going to be
replaced by PHP
On Tue, 5 May 2009, Abdul Basit wrote:
I wrote the php code for asterisk that was two page long and wasim baig sb
wrote the same stuff in 1/2 page line of code using python with
implementation of python libraries.
yeee!
If I wrote it in a single very long line of C would you be even
On Wed, May 6, 2009 at 1:51 AM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 5 May 2009, Abdul Basit wrote:
I wrote the php code for asterisk that was two page long and wasim baig
sb
wrote the same stuff in 1/2 page line of code using python with
implementation of python
This is a known bug. It is fixed in the trunk version of chan_mobile.
On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote:
Hello list,
I recently started testing the chan_mobile addon and after a
successful installation and configuration I have a couple of problems
that I can't fix
On Tue, 2009-05-05 at 12:41 -0700, Steve Edwards wrote:
AGI and AMI are only 2 tools at your disposal. In terms of ease of
implementation, I'd define the available set of tools and rank them in
this order when considering how to solve a problem:
) Dialplan
) pbx_lua
) AEL
) AGI
) AMI
Please elaborate; obviously ?? the dialplan is the simplest route to solve
any problem. Pbx_lua and ael are the next logical extensions to
dialplan(?), AGI can be simple custom code in PHP, Perl, C, etc. , AMI is
like AGI but using Manager interface and Asterisk Applications is tweaking,
On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Do you want to build your own?
If so, you can put togther a 1GHz fanless VIA miniITX board, case (that
will take a drive or flash IDE), memory and psu for well under £200. Same
system has one PCI slot
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Nicholson Sent: Tuesday, May 05, 2009 4:51 PM
On Tue, 2009-05-05 at 12:41 -0700, Steve Edwards wrote:
AGI and AMI are only 2 tools at your disposal. In terms of ease of
implementation, I'd define the available set of
I meant to scrub the files for my passwords but *failed* 0_0 and pwned my self.
Thanks for the link Jeff. Solved the problem... I was missing iodbc.
Thanks!
On Tue, May 5, 2009 at 10:27 AM, Tiago Durante tiagodura...@gmail.com wrote:
See if that helps you:
Thank you for your reply!
I downloaded the latest revision of asterisk trunk and asterisk-addons trunk
but it's not working at all, no key pulsation was detected. The last stable
release at least detects first key pulsation.
I checked out using:
svn checkout
Hi Folks,
I just wanted to share with you all some information about two
well-respected members of the OSS telephony community who will both be
speaking this year at ClueCon http://www.cluecon.com. Their topics are
relevant to Asterisk users so I felt compelled to let everyone know about
them.
Hi, all!
when my h323 phone dial in Asterisk system, i can hear nothing. and
the following is the log slice i picked from /var/log/asterisk/full.
ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1,
pwlib_v1_11_0, openh323_v1_19_0_1.
Best
Regards!
81948 [May 6 10:07:34]
I get a lot errors from chan_mobile when a call is in progress. More than
one line inserted every second.
ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9
Regards
On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz
carlos.ruizd...@gmail.comwrote:
Thank you for your reply!
I downloaded
Hi All,
Is there a way that I can bring the callerID obtained from an incoming call
to display in a web browser. I am using eyeBeam as softphone. Please help me
with this. Thanks in advance.
Regards,
Kurian Thayil.
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