Buggy module. I mis-read. and downloaded the one for pre 1.8.4
On Thu, Oct 18, 2012 at 12:53 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote:
3) im root
Glad to meet you.
:D
If you meant the user running Asterisk is root, this is a less than
optimal
situation that can lead to really big problems.
Why? Steve please explain.
Well, if an attacker
Resolved... Thanks to Digium for pointing out I installed a buggy module...
My tired eyes though 1.8.14 was 1.8.1.4 ... Good night all.
On Thu, Oct 18, 2012 at 12:35 AM, Jared Baxley jared.bax...@gmail.comwrote:
I was following Digium's instructions to the letter to install g729. but
upon
If you meant the user running Asterisk is root, this is a less than
optimal
situation that can lead to really big problems.
Why? Steve please explain.
Well, if an attacker manages to inject some code and Asterisk is running
as root, poof goes your system or you get an
On Wednesday 17 October 2012, bilal ghayyad wrote:
Actually I am not talking on how to handle it in the extensions.conf
because I am doing same as you wrote. But even so, I am facing a problem
that some calls are captured and some calls are not captured.
Currently, I set the callwaiting=no
Hi,
Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64
running Linux on 2012-03-08 23:05:09 UTC
We have some problem when running a AGI script (build with PHP), existing
channels (all of them) gets a hickup and then continues.
We are using AGI to lookup incoming calls
On Thursday 18 October 2012, Magnus Löfqvist wrote:
We have some problem when running a AGI script (build with PHP), existing
channels (all of them) gets a hickup and then continues. We are using
AGI to lookup incoming calls in directory.
It is kinda annoying, and I don't understand how it
Hi,
The script is not essentials in this.
I have tried with a empty script...
It stills generate problems with existing channels.
But if a try with a empty perl script, it dosent... seems to be something that
php is doing when starting up.
Best regards
Magnus
-Ursprungligt
On Thu, Oct 18, 2012 at 6:33 AM, Magnus Löfqvist m...@vmi.se wrote:
Hi,
** **
“Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a
x86_64 running Linux on 2012-03-08 23:05:09 UTC”
** **
We have some problem when running a AGI script (build with PHP), existing
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote:
any link for me so i can learn more about security practices with
asterisk? i'm using a public ip.
I think John Todd did a doc that is included with Asterisk.
http://nerdvittles.com/ has had a couple of recent articles on security.
There was a
On Thu, 18 Oct 2012, Magnus Löfqvist wrote:
I have tried with a empty script...
It stills generate problems with existing channels.
But if a try with a empty perl script, it dosent... seems to be
something that php is doing when starting up.
Can you explain more precisely what you mean by
Trying to set some CDR fields in the connected macro of a queue
command. None of the custom fields I set are stored in the database,
but I can set userfield and it does get set. I think that the macro
runs on the agent's channel, not the caller's, and this might contribute
to the problem.
I want to track the number of calls up at any given time, through the AMI. I
found the Link and Unlink commands as the most likely candidates - is that the
right way?
Also, a comment on the wiki suggests that Link may be called several times for
a single bridge if transcoding is required.
The simplest way to accurately do this would be to issue command core show
channels verbose
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 18, 2012 9:58 AM
To: Asterisk Users List
Subject:
I need to do this from the AMI (not the CLI)...I don't *think* a comparable
command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap a call
bridged,unbridged type event.
From: asterisk-users-boun...@lists.digium.com
The AMI Command function issues CLI commands, but carry on.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 18, 2012 10:20 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Counting calls
I guess you are looking for event handler, which can be polled
programatically n not via manual command entry?
Mitul
On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote:
The AMI Command function issues CLI commands, but carry on.
** **
*From:*
core show calls
works for me on asterisk-11
I know it's a poll, but it's a very simple call ;)
On 18 October 2012 16:28, Mitul Limbani mi...@enterux.in wrote:
I guess you are looking for event handler, which can be polled
programatically n not via manual command entry?
Mitul
On Oct 18,
Hi,
Though I've moved this functionality to freeSWITCH but here is the logic.
*AMI approach:*
Use perl POE and when received Bridged event increment the counter, and
similarly on Hangup decrement the counter. Since I see that there are some
issues mentioned by OP so I assume using plain
hi, I want to use asterisk as IVR system ,but to make and receive GSM call, i
want to use 3g usb modem.(voice
enabled)http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php
and i want to install this system on two different machine1 on mac os x -2
raspberry pi- (debian
Short answer is, its not possible
Long answer, why it is not !!
U would have to write a dahdi module for this 3G modem to help asterisk
understand it as standard gsm channel.
Hope that help,
Mitul
On Oct 18, 2012 9:16 PM, Mahendra Dobariya mahendra_mahen...@hotmail.com
wrote:
hi,
I want to
On 18 Oct 2012, at 17:19, Michelle Dupuis wrote:
I need to do this from the AMI (not the CLI)...I don't *think* a comparable
command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap a
call bridged,unbridged type event.
Michelle,
if you
I was wanting to call ChanIsAvail from AMI.
I logged in and issues command,
Action: Command
Command: ChanIsAvail DAHDI/1
my response was this:
event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR
][LF ]No such command 'ChanIsAvail DAHDI/1'
Is there any way to tell if a
Using AMI you might want to use hints instead of ChanIsAvail since that
command puts its' output back into the dialplan as variables.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday,
On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
U would have to write a dahdi module for this 3G modem to help asterisk
understand it as standard gsm channel.
Look up chan_datacard (i think that's what it's called from memory).
Steve
--
On 18 Oct 2012, at 17:58, Jerry Geis wrote:
I was wanting to call ChanIsAvail from AMI.
I logged in and issues command,
Action: Command
Command: ChanIsAvail DAHDI/1
my response was this:
event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF
]No such command
Considering this your real question I want to track the number of calls up
at any given time, through the AMI
Comparable Command Exists !!
You can simply run CLI commands *through *AMI and receive the response.
Command used for this is itself Command in AMI. If you still want to do
it by
Are there any developers that are familiar with the Queue() app implementation
and how it distributes calls?
Alex Forster
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New to Asterisk? Join us for a
Considering this your real question I want to track the number of
calls up at any given time, through the AMI
Comparable Command Exists !!
You can simply run CLI commands through AMI and receive the response.
Command used for this is itself Command in AMI. If you still want
to do
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote:
On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
U would have to write a dahdi module for this 3G modem to help
asterisk understand it as standard gsm channel.
Look up chan_datacard (i think that's what it's called from memory).
It seems fixing this issue and [1] would require significant efforts:
https://issues.asterisk.org/jira/browse/ASTERISK-20556
Too bad, it means DAHDI is a no way for me, I will have to switch to SIP
DECTs :(
Niccolò
It seems fixing this issue and [1] would require significant efforts:
https://issues.asterisk.org/jira/browse/ASTERISK-20556
Too bad, it means DAHDI is a no way for me, I will have to switch to SIP
DECTs :(
Niccolò
As you may know, asterisk version 10 and high use sqlite. Are any examples
or documentation how to use in dialplan?
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Hello!
I'm trying to use psi+ to conect to asterisk using chan_motif and vise
versa.
Connection looks good, but no sound.
As I see there is some traffic (22.229 is my desktop with psi)
08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82
08:38:37.481325 IP
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