Hi
I'm struggling to find the dependencies to allow me to tick
BETTER_BACKTRACES while installing asterisk 1.8.7 on CentOS 5.6
Does anyone know what I need to install to do this?
Regards
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
One of our clients has a Draytek Vigor 2920- their natted Snom phones
behind it are registered to an Asterisk 1.4 server on an external public IP.
I've set QOS, bandwidth management and turned off the SIP ALG via telnet
but I'm still having some problems with some of the phones losing
Hi list,*
*I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.*
Extensions.conf*
exten = 111,1,Answer()
same = n,Dial(SIP/2206,60,r)
same = n,Hangup()
*SIP.conf*
[2218]
type=friend
Hi John,
We've had similiar issues with customers behind the 2920 connecting to a hosted
asterisk system. If you rebooted a phone it often didn't re-register, Checking
the NAT sessions table on the router revealed stale nat sessions open for the
phone.
On the advice of Dreytek we found a fix
If the caller hangs up Asterisk sends a SIGHUP. You can catch the
signal and do whatever you want to do.
Am 21.11.2011 07:38, schrieb David Cunningham:
Hello,
We would like to continue a Perl AGI after a Dial() it has done
completes following caller
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the called
party to the specified context and extension and continue execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100,
As far as I know the linux kernel uses inotify to give Asterisk a hint,
that a new call file is available. Does inotify work in your environment
(external storage device) at all?
Am 18.11.2011 11:29, schrieb Ishfaq Malik:
We have a number of asterisk servers that share a spool directory on an
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will
get permission to try new firmware later!
JT
On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote:
Hi John,
We've had similiar issues with customers behind the 2920 connecting to a
hosted asterisk
On 11/20/2011 02:49 PM, Matt Hamilton wrote:
2. if the devices/members in the queue are not reachable, I would like
to forward him to a phone B.
I'm looking for a fast/practical way of accomplishing the second one.
In other words, before sending a call to a queue, I would like to see
if
Thorsten,
We have SIGHUP set to 'IGNORE', but it still does not continue the AGI
after the Dial(). Do you have any idea why that might happen?
Thanks for your advice.
On 21 November 2011 22:19, Thorsten Göllner t...@ovm-group.com wrote:
If the caller hangs up Asterisk sends a SIGHUP. You
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk wrote:
We do that with the F option in Dial().
From
hello,
try to delete all spaces between user and password on the pass.txt
Regards
- Bakko--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
hello list
i have asterisk 1.4 installed and i want to use CDR mysql during the
installation i didn’t check the cdr mysql with make menuselect
my question : i want to check this option now after the installtion and
configuration of all options but he asks me to do. /configure before to
Hi,
After deleting all space no improvements.
On Mon, Nov 21, 2011 at 5:35 PM, bakko asannu...@gmail.com wrote:
**
hello,
try to delete all spaces between user and password on the pass.txt
Regards
- Bakko
--
_
--
salaheddine elharit wrote:
because this server is very important for me and i can’t stop it
This is one of those schedule for after hours things then. I don't
believe you can do this without a restart of the Asterisk service. But,
down time should be minimal.
Doug
--
Ben Franklin
Hi,
I use an AGI with PHP. Here is a short snippet:
[...]
declare(ticks = 1);
pcntl_signal(SIGHUP, array($this, "signal_handler"));
[...]
public function signal_handler($signal_number)
{
$this-log_message("debug", "Signal catched:
On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
hello list
i have asterisk 1.4 installed and i want to use CDR mysql during the
installation i didn’t check the cdr mysql with make menuselect
my question : i want to check this option now after the
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to
avoid going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does
stuff - the AGI process just runs quickly then
I tried to patch app_read on my development dahdi box as follows:
static int unload_module(void)
{
int res;
res = ast_unregister_application(app);
/* ast_module_user_hangup_all(); */
return res;
}
But the offending behavior persists - it's not a show-stopper but it
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From:
i try to run make menuselect without configure but he give me an error and
he tell me that i must run ./configure before launch make menuselect
i'm afraid if i launch ./configure and after make menuselet to lost all
configuration related to asterisk
BTW i can restart asterisk without issue
From what I read you are running a pre-compiled asterisk - what you can do
in that instance is this
1 create a directory like /usr/local/src/asterisk/1.4-update
2 wget the matching version as indicated by core show version
3 extract the tar to the directory from step 1
4 run ./configure
5 run
Two items
#1 you only need 1 disallow=all in your sip.conf definition
#2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an xlite
response to Asterisk starting music-on-hold during the connect pause. The r
on the dial command attempts to do a faux ring which xlite interprets as a
Hi,
Does a parameter exist for a queue to delay ringing/sending a caller to all
agent phones after the previous call is answered by an agent? My queue ring
strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones.
And it looks like the KIRK wireless server may need a split
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Sunday, November 20, 2011 1:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Deleting AstDB family at start
Is it possible to delete the keys
hello list
i have asterisk 1.4 with and i have one card diguim (E1) with 2 providers
i have noticed by an error related to the first provider some times i can
not call the numbers of of this provider
but with the second one there is no issue alos i cal call the internal
extension without issue
thanks danny and thanks all i will test this solution and i will update you
by the result
kind regards
2011/11/21 Danny Nicholas da...@debsinc.com
From what I read you are running a pre-compiled asterisk – what you can
do in that instance is this
1 create a directory like
Hi,
Is there a way to add a uniqueid prefix to each server to make sure that the
CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL
tables to keep these records.
Regards,
Mike
--
_
--
In the dial plan language of asterisk, what is the difference between prompting
the user with a Playback() command vs. a Background() command? I want in a part
of my dial plan to ask the user a prompt, and wait for 4 digits to be typed in.
I don't want the user to have to end the string with a
So I found a good description of the timeoutrestart setting here
https://issues.asterisk.org/view.php?id=12690#87263. It definitely isn't what
I'm looking for. So I think I may be left with two options:
1. Set Skip Busy Agents to No. (not sure how this will work with my KIRK
phones. Currently
It sounds like you may want to use the READ command instead. This lets you
hard-set the number of digits to expect and then sets a variable which you
can use later in the dialplan. Generally you use the background command to
let them dial an extension or automated attendant option. Playback
First question - playback is not interruptable by DTMF, background is.
You have two options here
Option 1
Use Read
[getnum]
Exten = start,1,read(mydigit,prompt,4,skip,1,2)
.. verification stuff
Option 2
Use WaitExten with Background
[getnum]
Exten = start,1,background(prompt)
Exten =
Hi There,
I'm still having this problem, Does somebody know what can be
happening?
Regards.
On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
Hello,
The exten is the parameter passed to the macro, which contains the
sip device name. I'll change the name to another less
Yeah fastAGI is great, I've been using it for a while for performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem
Thanks Danny.
[clearkeys] Exten = start,1,answer() Exten = start,n,dbdeltree(foo)
Exten = start,n,hangup Set and retrieve Global variables for small
searches.
I will try the local call option to [clearkeys].
I guess I can also use a global flag to call dbdeltree only once in the
Since the MYSQL CDR is not the standard /var/log/asterisk/cdr-csv/Master.csv
file, but an add_on where uniqueid is just a table field varchar(32), you
could create an AGI to touch the field during the hangup extension and
append the servername or a number to the front, so instead of 123456.111 you
Mike,
Just enter a unique systemname into asterisk.conf for each box. This system
identifier is appended to the front of the unique id field in cdr.
/etc/asterisk/asterisk.conf
[options]
systemname=asterisk1
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 21 Nov
Thank you, just what I was looking for.
Danny: that`s a good solution, but I wanted something that didn't depend on
one more extra script running. I have plenty of those already.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Mon, 21 Nov 2011, Danny Nicholas wrote:
Option 2
Use WaitExten with Background
[getnum]
Exten = start,1,background(prompt)
Exten = start,n,waitexten(2)
Exten = ,1,noop(user pressed )
Exten = I,1,playback(invalid)
For option 2 you have to define each valid 4 digit entry in the
On Sun, 20 Nov 2011, Matt Hamilton wrote:
Is it possible to delete the keys belonging to a family in AstDB at
Asterisk startup? I would like to repopulate it from another source each
time Asterisk is restarted.
How about:
[sudo] /usr/sbin/asterisk -r -x 'database deltree example'
On 11-11-21 03:46 PM, Steve Edwards wrote:
On Sun, 20 Nov 2011, Matt Hamilton wrote:
Is it possible to delete the keys belonging to a family in AstDB at
Asterisk startup? I would like to repopulate it from another source
each time Asterisk is restarted.
How about:
[sudo] /usr/sbin/asterisk
What flavor does cli.conf start on?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Monday, November 21, 2011 3:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Hello again list,
I'm running a 1.4.42 install on SUSE with an
AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly
install 2.5 and also won't install Asterisk 10.0 because I can't get a good
SQLite3 library to install. Whenever I enter DTMF
Thanks Paul. The following works..
--cli.conf
---
[startup_commands]
;
; Any commands listed in this section will get automatically executed
; when Asterisk starts as a daemon or foreground process (-c).
;
;sip
Have you tried, instead of pre-processing the caller before calling
Queue(), checking the ${QUEUESTATUS} variable.
Even when the phones are UNREACHABLE, QUEUE is still trying until it times out
- ${QUEUESTATUS} = TIMEOUT
I get the following for all the members of the queue, in a loop,
The strange thing is that we are using fast AGI, and for some reason the
AGI always exits when the caller hangs up - even when I set HUP to IGNORE.
If I set HUP to a subroutine that just logs a message, that message is
never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley
Hi,
I'm trying to have Asterisk pick up a call and stream it to Liquidsoap
(Icecast2 compatible).
This is what I have in my extensions.conf :
[default]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Ices(/etc/asterisk/asterisk-ices.xml)
exten = s,n,HangUp
Here's what working so far:
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