Hi, sorry for my insistence but I would your aid for my problem.
Thanks.
--
Salvatore.
- Original Message -
From: Sasa s...@shoponweb.it
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 03, 2010 9:51 AM
Subject:
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.
--
Hi,
I'm developing an application using AMI and I need to get information
about incoming call _before_ queue member answers it.
I'm using static members (queue is pretty simple) and don't want to use
chan_agent (I think AgentCalled event will do what I'm looking for).
Here is what I'm getting
How annoying this is.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-09 3:59 AM, d...@accessgate.net wrote:
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net
Simple enough:
How can I get Asterisk version from within my dialplan? (preferably without
calling an AGI script that parses asterisk -rx show version)
Is it available as a global variable?
Vieri
--
_
-- Bandwidth
hi all,
I am calling a PSTN and trying to transfer it to another asterisk server
through exec_dial function.
$agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
Though this is the function written by me in a file inbound.php which is
called when an extension is dialled.
When ever
Let's all send John and Mary an email to tell them how thoughtful Doug is
and you can bet he will either turn off or modify his rule :-)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Wednesday, June
Since you have some inherent timeouts due to technology jumps, you might try
$agi-exec_dial(SIP,ww2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
To make asterisk wait 1 second before trying the call.
_
From: asterisk-users-boun...@lists.digium.com
Hi,
Is Astmanproxy still downloadable ?
At the moment, I can't download anything.
I'm usually using this http://github.com/davetroy/astmanproxy/tarball/masterURL
I can use a previous tar file but I would be pleased to know if I should do
something around this issue or not.
Regards
--
Starting with version 1.6.x there is a VERSION function that I think will give
you the version number.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 9, 2010, at 5:19 AM, Vieri wrote:
Simple enough:
How can I get Asterisk version from within my dialplan?
I use to use trixbox its basically asterisk with free pbx. What are your
extension numbers? Ring group number? What processor are you using? The more
info the better. When I used trixbox I never had this problem. It could be
DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind
Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to
be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it
can be used as a simple means to provide secure/encrypted calls.
GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch
GNU SIP
I have commented out case 5, case 2 and case 3, leaving case 1, 4,6,7,8,9.
But when I press 1 on the menu, I hear: I'm sorry, I did not
understand your response
if (play_auto) {
cmd = '1';
} else {
cmd = vm_intro(chan, vmu, vms);
}
Hi, here information request:
extension number is 100/101
ring group number is 600
cpu : Intel(R) Pentium(R) D CPU 3.00GHz 3 GHz
On another voip machine (always with Trixbox) I haven't this problem, I have
tried with another phones and with XLite I have always this problem.
About DTMF in SIP
On Wed, 9 Jun 2010, nikhil singhania wrote:
I am calling a PSTN and trying to transfer it to another asterisk server
through exec_dial function.
$agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
Though this is the function written by me in a file inbound.php which is
I checked out the sites and can't figure out what this thing is! (Without
delving into the documentation).
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew J. Roth
[mr...@imminc.com]
Sent:
On 08/06/2010 19:19, Steve Edwards wrote:
The ONLY way (how's that for humble) to do this in a reliable and robust
method is to use a real database. Personally, I like MySQL and I prefer to
do database work in an AGI in a compiled language like C.
Maintaining the accumulated duration in a
Good idea. I am emailing them right now.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-09 9:11 AM, Danny Nicholas da...@debsinc.com wrote:
Let’s all send John and Mary an email to tell them how thoughtful Doug is
and you can bet he will either turn off or modify his rule J
Dear all
i'm planning an upgrade of some asterisk installation from 1.4.32 to
1.6.0.28 (as i think it should be the most stable now).
Reading the UPGRADE-1.6.txt file i've noticed that:
* SIP: The call-limit option is marked as deprecated. It still works
in this version of
Asterisk, but will
Just is PRI line you can do it..
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
Date: Tue, 8 Jun
Dear all
after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt
file) the | delimiter is not working by default.
I've added a compat section in asterisk.conf a
[options]
dontwarn = yes
[compat]
pbx_realtime=1.4
res_agi=1.4
app_set=1.4
And restarted Asterisk, but i still have
Matthew J. Roth wrote:
Is anyone out there using SIP Witch in conjunction with Asterisk? It
claims to be able to enhance existing IP-PBX solutions such as
Asterisk, so maybe it can be used as a simple means to provide
secure/encrypted calls.
GNU SIP Witch - Summary
At 4:04 PM on 09 Jun 2010, Jonas Kellens wrote:
I have commented out case 5, case 2 and case 3, leaving case 1,
4,6,7,8,9.
But when I press 1 on the menu, I hear: I'm sorry, I did not
understand your response
Looks like someone broke the first rule of Optimization Club[1]. I
think you
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.
--
Hello users,
i am looking for a solution in terms of CDR for the outbound only call.
presently i have the following setup.
//extensions.conf
[from-outside]
exten = _X.,1,NoOp(IncomingCall)
exten = _X.,n,BackGround(choce.wav)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Hangup
exten =
Hello
I have also became like this problems and have found solution to make
outgoing calls via local channel, and now if my customer do a transfer,
I can calculate extra international outgoing calls.
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
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