Hi Paul,
Thanks a lot for the response.
I'm a novice so pardon me for the stupid questions. I thought that maybe the
PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM
it might be possible.
I basically want to know how Asterisk can dial out calls from the lines
Some thoughts inline:
logan wrote:
Hi Paul,
Thanks a lot for the response.
I'm a novice so pardon me for the stupid questions. I thought that maybe the
PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM
it might be possible.
I basically want to know how
hello,
why not use execif or gotoif?
this would look like this:
exten = _X.,n,ExecIf($[${EXTEN:${LEN(${EXTEN})-1}}=3]|do would ever
you want to do
best regards
steve
Vieri schrieb:
Hi,
How can I match an extension ending with 3 (just an example but applicable
to any other digit,
Thanks Paul. Your help is much appreciated here.
I don't really understand this question - Asterisk can make calls over
phone lines. And it does it well.
Surely, Asterisk does that well, but Asterisk needs to have multiple phone
lines for that. I thought that a traditional switchboard made
logan wrote:
Thanks Paul. Your help is much appreciated here.
No problem - been working on telephone systems for about 12 years now -
which doesn't even make me an old hand...
Surely, Asterisk does that well, but Asterisk needs to have multiple phone
lines for that. I thought that a
Hello, for a long time i am using asterisk 1.6 with astgui.
but for production system i intend to use asterisk 1.4 which i think might
be more robust. And for a more developed service options i preferd to
install with freepbx.
But still there are big plusses and minusses for both system.
My
can you try with elastix
it's same like freepbx but have some advance function like group exertions
and all those
On Mon, Jul 20, 2009 at 3:48 PM, Oguzhan Kayhan oguzh...@bilkent.edu.trwrote:
Hello, for a long time i am using asterisk 1.6 with astgui.
but for production system i intend to
Hello
I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to
send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2
How can I do that load balancing in extensions.conf?
I have something like this:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
; 2 in 3
Take a look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
You should be able to do what you want with this, it obviously won't take in
to account the actual amount of people still in the queue (for example if
someone hangs up while on hold). I'm sure there'd be a way of
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten = 123,11,Hangup
Joao Pereira
--
StarTel - A Rede Livre
Joao
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten =
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
exten = 123,6,Queue(queue_1)
exten = 123,7,Hangup
exten =
Danny Nicholas schrieb:
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
...,Set(CALLCOUNT=$[${CALLCOUNT} + 1])
or
...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)})
exten =
Tim, this is a partial solution. The find as written would remove
greetings, unavailable messages, etc. You would need to add a grep to get
only msg files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim
Tilghman Lesher wrote:
My understanding of QUEUE_MEMBER_COUNT just give a total of agents in the
queue.
Synopsis Count number of members answering a queue
It may or may not be the answer to the OP's question, depending upon what he
meant by available. Without clarification, it's
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding
Hi Gang,
I've got the latest SVN branch of 1.4 downloaded onto SUSE
11.0. Everything is happy EXCEPT, I can't get fax to be recognized by make
menuselect. I tried copying app_rxfax.c and app_txfax.c to the apps
directory and starting again from ./configure, but no joy. Any
Hello -
I've been running Asterisk (quite happily!) for several years now
using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM).
I'm also running another old PC running m0n0wall as a firewall.
Between these two boxes, that run 24x7, I'm drawing a lot more power
than needed and
On Sat, Jul 18, 2009 at 12:05:51PM -0400, Jerry Geis wrote:
/ I am current running on a production system
// zaptel 1.4.12.1
// libxpri 1.4.1
// asterisk 1.4.25
//
// The above configuration works.
//
// I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25
// This did
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, July 17, 2009 5:33 PM
Write up a small shell script that uses 'find
/var/spool/asterisk/voicemail/ -mtime +2' for a list of files older than
two days assuming you want ALL files
Look into AstLinux as one possible solution for both Asterisk and a
firewall on the 5501, with no hard drive.
John Novack
Brian McEntire wrote:
Hello -
I've been running Asterisk (quite happily!) for several years now
using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM).
Emrah wrote:
This is an asterisk-users question, and would have been more appropriate to
have
asked there.
Instead of setting up your conferences in meetme.conf, you could set them up
dynamically in the dialplan, and then you can control whether the user is
prompted for a pin or not
I am using an IBM Server, after while in the MBR it said that Event logs are
full, so after clearing it, the asterisk can't run.
i think it deleted a file, so which file i have to create again. and what's its
chmod.
Thanks
_
With
Probably /var/log/asterisk/messages 0644.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T
Sent: Monday, July 20, 2009 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Event Log
I am
At 10:09 AM 7/20/2009, you wrote:
If the Soekris isn't expected to work well, are there any mainstream
small form factor/low-power solutions for a SoHo asterisk server?
I just built a box for my Asterisk system using an Intel Motherboard
with an Atom 330, 5400 RPM HD, TDM 400 with 4 red cards
Hi Nicholas!
Perhaps, there are other ways as I describe here, but I use this way
successfully about 4 years
- install latest spandsp version
- went to root directory of your svn asterisk
- type make distclean (because there are preconfigured things in
downloaded version)
- change to
On Mon, 20 Jul 2009, Danny Nicholas wrote:
Any suggestions?
Sorry. I can't resist :)
Asking a question with a useless Subject.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
No, this file is still existed,
i think it's another file.
Thanks
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 20 Jul 2009 13:20:23 -0500
Subject: Re: [asterisk-users] Event Log
Probably /var/log/asterisk/messages 0644.
From:
Asterisk -vc should tell you what it wants to be able to start.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T
Sent: Monday, July 20, 2009 1:41 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Thanks Ira -
I may yet still go with a standard Intel solution, but I think there
could be major power savings to be had going with a smaller box like a
Soekris if it can work. A good rule of thumb for 24x7 devices is $1
per watt per year, so 45 watts, while good, will still be $45 per
year. I
At 12:47 PM 7/20/2009, you wrote:
I may yet still go with a
standard Intel solution, but I think there
could be major power savings to be had going with a smaller box like
a
Soekris if it can work. A good rule of thumb for 24x7 devices is $1
per watt per year, so 45 watts, while good, will still
On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote:
Hello -
I've been running Asterisk (quite happily!) for several years now
using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM).
I'm also running another old PC running m0n0wall as a firewall.
Between these two
Should SipPhone support ISN routing for their 747 ITAD? Cast a vote:
http://forums.gizmo5.com/viewtopic.php?t=10197
Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD
#1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever
(Karl tips his hat to Ward
Thanks for the reply Alex. I'm not too scared of the soldering iron (I
own one, but my work with it isn't pretty ;-)
But can you confirm, are you just using the small power header on the
board to supply power to the pci card? I was wondering if I was going
to have to snake an another wall wort
Hi,
I have an extension which I want to use only for x-lite, and don't want
anybody to register IP phones on it. I can see that 'sip show peer 3547'
shows softphone's id. Is there a way to restrict registrations on this
extension by useragent id?
I googled but so far couldn't find any way to do
Have you solved this issue?
When I restart the machines I can't make an outgoing DAHDI call until I get
an incoming call on that same line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: viernes,
On about 25% of inbound calls to a ring group, picking up any one
extension as it rings results in dead air.
Some details regarding my VoIP network to make the following logs more
readable:
192.168.7.130 resolves to the trixbox host.
192.168.7.135 resolves to endpoint 812.
192.168.7.137
On Mon, 20 Jul 2009, Brian McEntire wrote:
A good rule of thumb for 24x7 devices is $1 per watt per year, so 45
watts, while good, will still be $45 per year.
In San Diego, CA we pay $0.33 per kWh (Over 200% of Baseline rate). With
8,760 hours in a year. That works out to $2.98 per watt or
I still don't see what you gain by using m0n0wall and a separate
Asterisk install. I can't think of one thing that you would need a
separate m0n0wall instance to do that AstLinux can't do on it's own.
The web interface has become quite completely in the last few releases.
Traffic shaping,
--
I know it doesn't really sound very helpful to blame the entire server
manufacturer, but some others might agree, brand spanking new and shiny
might not be the best thing for Asterisk, especially these cards.
There's nothing wrong with brand spanking new and
Sadly, at the end of the day the answers will probably be no, no, no and no.
PaulH
logan wrote:
Hi,
I'm an absolute newbie and wanted to know the following.
I want to have a setup where I have a PSTN line connected to my
Asterisk box and want to know if it is possible to make more than
Darrick -
You seem adamant, and I will look deeper into the firewall in Astlinux! :-)
The one thing running monowall in a VM would do for me is (in theory)
make it very simple to move my existing, working m0n0wall
configuration. I've been running it for a while, it serves a bunch of
DHCP
Sebastian wrote:
Have you solved this issue?
When I restart the machines I can't make an outgoing DAHDI call until I get
an incoming call on that same line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Brian McEntire wrote:
Darrick -
You seem adamant, and I will look deeper into the firewall in Astlinux! :-)
Brian,
I am one of the developers, so I happen to like what we've done. There
have been some huge changes to the web interface and the overall project
in the past year or so.
At 03:30 PM 7/20/2009, you wrote:
Have you solved this issue?
When I restart the machines I can't make an outgoing DAHDI call until I get
an incoming call on that same line.
I've not and from the responses it's sounds like a known problem
Ira
___
hello:
I wan to use the test tools-patgen and pattest for pri cards. according to
Tzafrir Cohen from
http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know how to
use that.
do i need to connect two pri cards with two servers, or use a cable to
connect two cards in one server?
*I have the following scenario :
**
** kamailio User(222) --- Asterisk GW
Call to kamailio user
**--1234786
**
** |
**
** |
**
**
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