Hi,
Installation of dahdi requires kernel source that is not available with my
remote virtual machine. Therefore I installed Asterisk without installing dahdi
but when I start Asterisk it crashes while loading chan_agent.so (noload is
also not useful in this case).
Any suggestions or hints to
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.
These releases are available for immediate download at
http://downloads.asterisk.org
Asterisk Project Security Advisory - AST-2011-006
ProductAsterisk
SummaryAsterisk Manager User Shell Access
Nature of Advisory Permission Escalation
Asterisk Project Security Advisory - AST-2011-005
Product Asterisk
Summary File Descriptor Resource Exhaustion
Nature of Advisory Denial of Service
On Thu, Apr 21, 2011 at 4:30 PM, Steve Edwards
wrote:
> On Thu, 21 Apr 2011, Mark Deneen wrote:
>
>> I use runit to manage the asterisk process, and the chpst program
>> allows fine control over environment and other limits.
>
> runit is intended to be a sysvinit (/sbin/init) replacement and is no
On Thu, 21 Apr 2011, Mark Deneen wrote:
I use runit to manage the asterisk process, and the chpst program
allows fine control over environment and other limits.
runit is intended to be a sysvinit (/sbin/init) replacement and is not
installed (by default) on CentOS or Ubuntu distributions.
C
On Thu, Apr 21, 2011 at 3:23 PM, Steve Edwards
wrote:
> On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote:
>
>> Can't get hostname environment variable on asterisk dialplan.
>
> 1) Is HOSTNAME in the Asterisk process's environment? What does executing:
>
> tr '\000' '\n' < /proc/$(cat /var/run/ast
On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote:
Can't get hostname environment variable on asterisk dialplan.
1) Is HOSTNAME in the Asterisk process's environment? What does executing:
tr '\000' '\n' < /proc/$(cat /var/run/asterisk.pid)/environ
show on the shell console?
2) What does ex
Asterisk 1.8.4-rc2 (and 1.8.3)
DAHDI Version: 2.4.1.2
libpri version: 1.4.12-beta3
We are having a problem with getting the nationalprefix option of
chan_dahdi.conf to work. National calls do not have a "1" added to them when
nationalprefix=1. The PRI debug shows the call coming in as a Nation
Rajnikant,
This surely depends on how you start asterisk. How are you starting
the asterisk process?
-M
On Thu, Apr 21, 2011 at 7:20 AM, RAJNIKANT VANZA wrote:
> Hi Friend,
>
> Can't get hostname environment variable on asterisk dialplan.
> Help me about how to get hostname environment variabl
I am always googleing before putting anything here.. I was confused that's why
i came across to you guys! Still i am confused :(
-S
Date: Thu, 21 Apr 2011 13:01:52 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notificati
On Thu, Apr 21, 2011 at 12:26 PM, satish patel wrote:
> Hi,
>
> I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed
> call notification but i am having issue. following is my dialplan
>
> [macro-stdexten]
> exten => s,1,Dial(${ARG2})
> exten => s,2,Goto(s-${DIALSTATUS},1)
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call
notification but i am having issue. following is my dialplan
[macro-stdexten]
exten => s,1,Dial(${ARG2})
exten => s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANU
Hi,
Can anyone let me know how I can enable video (h.263) on SIP, but if a
video call is passed over IAX, it will remove the video and pass the
audio only.
What I tried was:
SIP - videosupport=yes
- disallow=all
- allow=alaw
- allow=h263
IAX - disallow=all
- allow=alaw
Might be both end codace are different. That's why this message is come.
On Thu, Apr 21, 2011 at 7:26 PM, Danny Nicholas wrote:
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Eric Wieling
> > Sen
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Thursday, April 21, 2011 8:54 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Transcode u
We are getting the following on 1.8.3 and 1.8.4-rc2, HELP!
Why is Asterisk unable to transcode to/from ulaw and g722?
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to
transmit frame type ulaw, while native formats is 0x1000 (g722) read/write =
0x1000 (g722)/0x1000 (g722
On 04/21/2011 08:12 PM, Khaled W. Chehab wrote:
Dears,
I configured an account on my asterisk pbx to record the outgoing calls.
When the asterisk pbx user make a call and send a fax the call
recorded to wave file format.
I searched the internet and found a software that can play the
reco
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Paul van der Vlis
> Sent: Thursday, April 21, 2011 3:09 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] allowguest=yes, how?
>
> Op
On 21 Apr 2011, at 13:46, A J Stiles wrote:
> You *might* be able to recover the document, *if and only if* the recording
> quality is high enough. Easiest way to try it is to call up a fax machine
> (either an actual real one, or a copy of Hylafax) from Asterisk and play the
> wav file down
On Thursday 21 Apr 2011, Khaled W. Chehab wrote:
> Dears,
>
>
>
> I configured an account on my asterisk pbx to record the outgoing calls.
>
> When the asterisk pbx user make a call and send a fax the call recorded to
> wave file format.
>
> I searched the internet and found a software that can p
On Wed, Apr 20, 2011 at 4:10 PM, Mark Deneen wrote:
> On Wed, Apr 20, 2011 at 4:35 PM, satish patel
> wrote:
> >
> > Hey Thanks for that reply after add following option it works but the
> text
> > output is totally different.. what its totally different is this
> dictionary
> > problem ?
> >
>
On Thu, Apr 21, 2011 at 4:03 AM, DHAVAL INDRODIYA
wrote:
> Hi,
>
> You can use
>
> Meetme(1234,dL(1800))
>
> where 1800 = 6 hours after 6 hours channel is hanf up
>
> regards
> Dhaval
>
>
>
> On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman wrote:
>
>> Is there a way to place a hangup tim
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname environment variable on asterisk dialplan.
I have written "export HOSTNAME" in /root/.bash_profile and when i execute
"echo $HOSTNAME" then get right hostname but not success through asteri
Hi all,
I have that when voice mail is forward to any guys then some text will
include into recorded file like
"this is a forward voice mail from extension or Mr. Virendra" the actual
file will play
Is it possible ? if yes then how ?
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Hi,
You can use
Meetme(1234,dL(1800))
where 1800 = 6 hours after 6 hours channel is hanf up
regards
Dhaval
On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman wrote:
> Is there a way to place a hangup time on a dynamic Meetme conference. I am
> using Page() with a Meetme conf and I ha
Hi Jose
thank you for your help after this command "yum install asterisk16-configs*
"all is ok now i can install asterisk without problem
Kind Regards
Salah.
2011/4/20 Jose P. Espinal
> but when i finished i found a file in etc asterisk but is empty
>>
>>
>
> You found "a file" in "etc aste
Op 20-04-11 21:47, Danny Nicholas schreef:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Paul van der Vlis
>> Sent: Wednesday, April 20, 2011 2:41 PM
>> To: asterisk-users@lists.digium.com
>> Subject:
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