Hi,
I don't use a macro.
I stay in the same dialplan (application)
In the h exten I place a test (for example testThis is a test/test)
If I look at the CLI and after I placed the example text in the variable
CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield).
But if
Hi
i want add a numeric password to a call in :
User call to a number,
Asterisk answer and request: please insert your pin code
the user enter a numeric code of 4 number and #
when asterisk have the code, he start a api.
Anyone have a sample of extension.conf for this ?
thanks
Olivier
--
Hi,
If I try to call out with Queue mechanism and the call is answered then hangup,
the CDR(userfield) in the h exten is placed in the CDR.
So for now I see that this problem only occurs with a Dial in the dialplan.
--
Arjan Kroon
-Oorspronkelijk bericht-
Van:
Did you take a look at
/var/log/syslog
/var/log/asterisk/messages
?
Using Debian? Take a look at iotop (apt-get install iotop). There you
can see information about which process consumes high io load.
Am 04.04.2011 17:23, schrieb Maximilian Grobecker:
Hello Thorsten,
the system has 4 GB RAM
Hi,
New update.
When I use the option g in a dial then the CDR fields are not updated.
When I perform a dial without the option g, for example rR then the CDR field
will be written in the h exten.
So therefore I lose the g option in the dial.
--
Arjan Kroon
-Oorspronkelijk bericht-
OK Dears;
Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the
equivalent?
I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if
someone can advise me:
Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, )
in new stack
[Apr
Change Wait,2 to wait(2)
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 5 Apr 2011 01:31:11
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Mensaje original-
Olivier CALVANO
Enviado el: martes, 05 de abril de 2011 9:16
Hi
i want add a numeric password to a call in :
User call to a number,
Asterisk answer and request: please insert your pin code
the user enter a numeric code of 4 number and #
when asterisk have the
Also change DeadAGI,a2billing.php to AGI(a2billing.php)
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 5 Apr 2011 01:31:11
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
I am using asterisk 1.4.2 and it usually does enforce the limit. yesterday
and couple of times before was an exception. I am still trying to find the
reason behind. Any more suggestions please?
oh by the way * 1.8.1.1 does enforce the call limit, i tested it yesteday on
sip channels.
On Mon, Apr
Jerry Geis wrote:
I am calling from a polycom phone into asterisk ( 1105 ) on a PC with
a speaker attached.
When asterisk first starts this works. In fact it works for some time.
Then it just stops with this error on the CLI.
[Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358
Hi,
the log files contained (sometimes) lines about refcount -1 in astobj.c.
I also generated core dumps and analyzed them - but there were always
errors in another module.
Mabye I found the solution:
Asterisk seems to crash when a required module cannot be loaded fast
enough due to heavy disk
Idea:
If something is corrupting your dialplan, then this should
reveal the extent of the corruption:
You might, when the system is working properly, do a:
asterisk -rx dialplan show somefile1
and then, when you are having problems, do a:
asterisk -rx dialplan show somefile2
diff -u
Steve Murphy wrote:
Idea:
If something is corrupting your dialplan, then this should
reveal the extent of the corruption:
You might, when the system is working properly, do a:
asterisk -rx dialplan show somefile1
and then, when you are having problems, do a:
asterisk -rx dialplan show
Jerry Geis wrote:
Steve Murphy wrote:
Idea:
If something is corrupting your dialplan, then this should
reveal the extent of the corruption:
You might, when the system is working properly, do a:
asterisk -rx dialplan show somefile1
and then, when you are having problems, do a:
asterisk
Hello
I'm no expert of iptables, and it seems like it can handle banning
IP's that are trying to register and fail too many times.
I'd like to use this feature instead of having to install a second
tool such as SSHGuard or BFS that parses the logs and reconfigure
iptables on the fly.
Is
Hi, I installed the Vestec module to one of my development Asterisk
servers a few months ago but now I need to move the license to another
host. Does anyone know how to do this? I've had a look on my Account
page on the Digium website but it only shows the Language Pack, and I
can't do anything
If you purchased from Diguim, call their tech support. If you purchased it
from Vestec, you'll have to provide them with some paperwork or shell out
some bucks.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Talk to Vestec.
As far as i know they are they ones that can re-issue the license code.
Kashif Kahn (kahn at vestec.com) was very helpfull whenever i need it
info for my project.
Stelios
On Tue, 2011-04-05 at 15:36 +0100, Lee Archer wrote:
Hi, I installed the Vestec module to one of my
On Tue, 5 Apr 2011, Gilles wrote:
I'm no expert of iptables, and it seems like it can handle banning
IP's that are trying to register and fail too many times.
Is there a good iptables configuration that I could use as reference?
Gordon Henderson posted a link to his script that handled
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing
Oh, you are *not* going to like this, but
you have a few different paths:
1. If the dialplan stuff is not really a memory corruption, but some sort of
unplanned,
but maybe accidentally programmed thing, either by you or something in
the asterisk
code, then:
a. compile asterisk for
Nevermind,
I have solved it my self. this script wring some logs in /tmp and somehow
logfile was already there. so just deleted and it works!
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 16:35:37 +
Subject: [asterisk-users] allpage issu on
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.
For instance, a caller leaves a voicemail, the voicemail will then call the
owner of the voicemailbox determined by a database look up.
--
_
Hi Rizwan
Thank you for your help i will test this solution and i will update you as
soon as i have any result.
Kind Regards
2011/4/4 Rizwan Hisham rizwanhas...@gmail.com
Do this:
exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)
you can also use the dial command for this as
Is it possible to create a voicemail box using AGI? How does asterisk know
about mailboxes when using Asterisk with pure AGI?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
vip killa wrote:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI
I don't have an AGI, but I do have dial-plan code.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
On Tue, 5 Apr 2011, vip killa wrote:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.For instance, a caller leaves a voicemail, the
voicemail will then call the owner of the voicemailbox determined by a
database look up.
Use 'mailcmd' in
On Tue, 5 Apr 2011, vip killa wrote:
Is it possible to create a voicemail box using AGI?
An AGI executes as a child process when a channel executes agi() via the
dialplan.
Are you intending to call into Asterisk and let the caller create
mailboxes?
All the AGI needs to do is add a line
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk hints
Hey guys!
I am new in hints application. what is the use of
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
From:
On my Polycom 501's I use hints to populate a buddy list - I hit the
buddies softkey and can see if my buddy is on the line.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 1:19 PM
fail2ban might be good for this.
On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote:
Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT)
From: Steve Edwardsasterisk@sedwards.com
Subject: Re: [asterisk-users] Iptables configuration to handle brute
force registrations?
On
If i want to watch every phone status Idel or Inuse the how should i use hint
in my dialplan. I meant should i need to specify each and every extension ? or
is there any catch-all extensions ?
-Satish
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 13:20:45
On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelson b...@cosi.com wrote:
fail2ban might be good for this.
I think you missed the point, which is reducing the need for an external
application that searches logs in order to determine whether or not to block
an IP.
Why run fail2ban and add overhead
Under linux-2.6.38 I was able to compile and install dahdi, however when
I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have
an old 400P card with one FXS and one FXO module. I have
dahdi-trunk r9868 and dahdi-tools-trunk 8670.
How can I get this to work correctly?
Thanks in
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 1:53 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dahdi and linux-2.6.38
Under
On 04/05/2011 01:52 PM, cov...@ccs.covici.com wrote:
Under linux-2.6.38 I was able to compile and install dahdi, however when
I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have
an old 400P card with one FXS and one FXO module. I have
dahdi-trunk r9868 and dahdi-tools-trunk
On 04/05/2011 01:54 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 1:53 PM
To: asterisk-users@lists.digium.com
Subject:
On 04/05/2011 02:00 PM, Shaun Ruffell wrote:
Would it be possible to update your email client to add a prefix to
quoted lines? It's hard for me to see where the message you're replying
to ends and your reply begins.
Arghh. I meant that to be a private email. My apologies.
--
Shaun
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, April 05, 2011 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dahdi and
On Tue, 5 Apr 2011, Sherwood McGowan wrote:
Why run fail2ban and add overhead when you can just do the same thing
with iptables itself?
Because it's not the same?
The iptables approach is great because it is 'light-weight' and it should
already 'be there.' Also, it can react quicker because
On 4/5/2011 2:11 PM, Steve Edwards wrote:
On Tue, 5 Apr 2011, Sherwood McGowan wrote:
Why run fail2ban and add overhead when you can just do the same thing
with iptables itself?
Because it's not the same?
The iptables approach is great because it is 'light-weight' and it
should already
Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 1:53 PM
To: asterisk-users@lists.digium.com
Subject:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dahdi and
Shaun Ruffell sruff...@digium.com wrote:
On 04/05/2011 01:52 PM, cov...@ccs.covici.com wrote:
Under linux-2.6.38 I was able to compile and install dahdi, however when
I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have
an old 400P card with one FXS and one FXO module. I
On Tue, Apr 05, 2011 at 02:03:20PM -0500, Danny Nicholas wrote:
On 04/05/2011 02:00 PM, Shaun Ruffell wrote:
Would it be possible to update your email client to add a prefix to
quoted lines? It's hard for me to see where the message you're
replying to ends and your reply begins.
Will see
On Tue, 5 Apr 2011, Sherwood McGowan wrote:
Why run fail2ban and add overhead when you can just do the same thing
with iptables itself?
On 4/5/2011 2:11 PM, Steve Edwards wrote:
Because it's not the same?
The iptables approach is great because it is 'light-weight' and it
should
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, April 05, 2011 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dahdi and
On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote:
Message: 12
Date: Tue, 5 Apr 2011 13:36:21 -0500
From: Sherwood McGowansherwood.mcgo...@gmail.com
Subject: Re: [asterisk-users] Iptables configuration to handle brute,
force registrations?
To: Asterisk Users Mailing
On Tue, Apr 05, 2011 at 02:43:15PM -0500, Danny Nicholas wrote:
On 04/05/2011 02:00 PM, Shaun Ruffell wrote:
On Tue, Apr 05, 2011 at 02:03:20PM -0500, Danny Nicholas wrote:
On 04/05/2011 02:00 PM, Shaun Ruffell wrote:
Would it be possible to update your email client to add a prefix to
quoted
Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial
On 04/05/2011 02:26 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Tuesday, April 05, 2011 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Tuesday, April 05, 2011 3:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dahdi and linux-2.6.38
On
Ok thanks I found the problem
The spa8000 has some bugs with t38 which are fixed in the spa2102 but not in
the 8000
1. If the adapter starts with g711 It doesn't switch to t38
2. (This my problem) when it does go to t38 and the itsp asks for it to
fallback to 9600 it doesn't fallback so
So I get this warning:
app_voicemail.c:11084 load_config: maxsilence should be less than
minmessage or you may get empty messages
Can anyone explain that a little better? When would I end up getting
empty messages if say minmessage was set to 3 seconds and maxsilence is
set to 10? 10 seconds
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
--
On Tue, Apr 5, 2011 at 2:40 PM, Steve Edwards asterisk@sedwards.comwrote:
snip
Are there possibly other drawbacks that I'm not seeing/remembering? I've
been running an iptables based setup for some time, never really jumped into
the fail2ban wagon
I've never used fail2ban either. I
First, this appears to be working for me though I'm not 100% sure of
that and cannot guarantee it will for you in any way, shape or form.
With the lawyering out of the way...
I've seen fail2ban allow more than 500 failed SIP login attempts in
under 30 seconds before adding an iptables rule to
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Paul Dugas
Sent: Tuesday, April 05, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Iptables
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I am seeing random crashes. For some reason the builds lock up and
stop taking sip connections. Existing calls stay on but when the user hangs
up no new calls or reg attempts work. In most cases a core restart now
On 6/04/11 12:39 AM, Maximilian Grobecker wrote:
Hi,
the log files contained (sometimes) lines about refcount -1 in astobj.c.
I also generated core dumps and analyzed them - but there were always
errors in another module.
Mabye I found the solution:
Asterisk seems to crash when a required
On Tuesday 05 April 2011 20:10:48 Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I am seeing random crashes. For some reason the builds lock up
and stop taking sip connections. Existing calls stay on but when the
user hangs up no new
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