Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two of these modifications are:
1- A proprietary configuration driver that will communicate with a
server that will be the
this variable, please let me know.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option
Hello there!
The only available way to control call duration is using the RTCC patch
(discussed here https://issues.asterisk.org/view.php?id=6335; and
mainteined here http://ast.varna.net/;) ?
The purpouse is to have a way to monitor (probably on a per-minute
basis) and hangup costly calls
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
Thanks a lot Faheem for you help.
I totaly understand now the approach you've used.
It's very interesting and inventive for sure.
I didn't know that I could append IP:Port info on user when using the
Dial command and that this will make calling to two different devices
registered using same
Thank you very much for all your help, Muhammad! (please let me know if
I should call you Faheem, instead).
I'll make some tests with this script on my premises as soon as possible.
Having a look on it, I couldn't realize how it really works in
conjunction with Asterisk.
I mean, it seems that
varchars(30)
Please adjust the table fields appropriately.
Hope this code block will solve you problems.
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com
--- On *Fri, 8/28/09, Mauro Sergio Ferreira Brasil
...@10.0.0.150:6060
The complete script is attached.
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com http://advcomm.net/
--- On *Wed, 8/26/09, Mauro Sergio Ferreira Brasil
/mauro.bra...@tqi.com.br/* wrote
Thanks Atis, its working pretty fine now.
Best regards,
Mauro.
Atis Lezdins escreveu:
On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira
Brasilmauro.bra...@tqi.com.br wrote:
Hello there!
Problem found.
For some reason, the update statement below is generated with an invalid
Hello there!
We are planning to use Asterisk on our VoIP platform, and we are
spending some brains on a way to provide the following facility: let
some SIP user (extension) registrate with more than one client (ATA,
SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate
calls
Hi Elliot, and thanks for the reply.
I'm not completely sure you've considered that the SIP users registered
on all devices are the same.
Have you ?
I mean...
How will I use Dial command with a sequence of same devices, like:
Dial(SIP/101SIP/101SIP/101), for example ?
That's why we are
Hi Barry, and thanks for the reply!
This was the first question I've made on meeting yesterday to decide
about this facility.
Having me here today making this question should give you an idea of the
level of acceptance of my suggestion :-).
Anyway, the idea is really try to make it work with
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent: Wednesday, August 26, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple user registration
Thanks again Barry for the help and attention.
Thanks for wishing me lucky as well... If we insist on this road I'll
need it for sure :-).
I can't agree more with your position, and I'll try to be sure our
commercial demands can't be acchieved with normal approaches before
adventuring on such
Hello there!
I was testing Asterisk for the last two weeks using the Realtime driver
for MySQL, and leaving rtcachefriends=yes configured to enable MWI.
Today I started making additional tests with rtcachefriends=no because
we will probably need to use Asterisk without this cache.
For some
any idea ?
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I was testing Asterisk for the last two weeks using the Realtime
driver for MySQL, and leaving rtcachefriends=yes configured to
enable MWI.
Today I started making additional tests
Hello there!
I need some help to configure two Asterix boxes to route calls using SIP.
I followed the instructions present at this site:
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;,
but I couldn't get it working so far.
The only
Hi guys!
The problem was solved by the use of same password for registration
users of both boxes.
Is there no way to indicate different password for registration user of
Box1 and registration user of Box2 ?
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello
Hello there!
During some research on Internet I found the following comparison on
site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;):
The main points listed on Asterisk's CONS that concerned me were:
* Conferencing on Asterisk depends on Zaptel hardware and/or kernel
going to use Asterisk for. Sounds like it is
for conferencing. Would you care to elaborate?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent: Tuesday, August 18, 2009
21 matches
Mail list logo
