-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Friday, April 4, 2014 11:15 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around the country and
connect from different network, so
real users
are located.
-=Michelle=-
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
Sent: Friday, April 4, 2014 11:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
Hello
I don't know what platform you are on, but if you are on Linux (and
possibly BSD) you could use fail2ban to block them at the network
interface.
On 04/04/2014 09:00 AM, motty cruz wrote:
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]:
On 4 April 2014 15:00, motty cruz motty.c...@gmail.com wrote:
Hello All, my asterisk server is constantly under attack
Unfortunately you are not alone.
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for
On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.com wrote:
I don't know what platform you are on, but if you are on Linux (and
possibly BSD) you could use fail2ban to block them at the network
interface.
I second fail2ban. If you need some ideas to configure it, you
thank you all for your support. I am using Linux, I only have about 7 users
outside our home network. I will learn fail2ban and will use it
accordingly.
again Thanks for your support.
On Fri, Apr 4, 2014 at 7:09 AM, Mauricio Tavares raubvo...@gmail.comwrote:
On Fri, Apr 4, 2014 at 10:05
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa), detection of break-in patterns
(even if someone guessed your un/pw), detect changes in dial rates, etc.
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa), detection of break-in
patterns (even if someone guessed
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa
absolutely right A J, thanks for the heads up.
I do not intent to implement that solution in production server, I hope to
learn it first, build a test server and monitor for a few days or weeks.
Thanks again,
On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On
On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:
thank you all for your support. I am using Linux, I only have about 7
users outside our home network. I will learn fail2ban and will use it
accordingly.
again Thanks for your support.
Do the 7 users outside of your home
Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.
Thanks for your support.
On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 4 April 2014 15:22, motty cruz
Well in that case fail2ban gets my vote.
On 4 April 2014 16:15, motty cruz motty.c...@gmail.com wrote:
Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.
Thanks for your support.
On
: Friday, April 4, 2014 11:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around the country and connect from
different network, so it won't be possible to lock it down to specific IP.
Thanks for your support.
On Fri, Apr 4
Use allowguest=no
And define ACLs for every SIP account.
And obviously, fail2ban for blocking suspicious IPs.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
:* asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Friday, April 4, 2014 11:15 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around
-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Friday, April 04, 2014 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6
that sounds feasible, Thanks Michelle,
On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello Steve,
After checking, I confirm that the call is cut precisely to 900 seconds
(15 min).
10.4.0.1 =
Can u debug on AS ?
On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello
Le 7/03/13 11:12, Mickael Monsieur a écrit :
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back
Update:
No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which
worked with no problem.
Probably if I have some time, I will do more testing with version 1.8.7 to
see what the difference is and what changes need to be made for this kind
of setup to work in 1.8.7
Joseph
On Mon,
On Sat, 16 Jul 2011, Zarko Zivanovic wrote:
I am sure that someone can help with this. We decided to do a fresh
install of asterisk 1.6.2.19 And after we did that, the problem that we
have is this – We cant run a single Php file!
testera.agi: Failed to execute
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
A couple things -
First, in extensions.con your context is [my-phone], but you're using my-phones
in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in order
for asterisk to answer the incoming analog call.
Third, I think you've got some
Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi
: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
A couple things -
First, in extensions.con your context is [my-phone], but you're using
my-phones in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in
order for asterisk to answer the incoming
On Sun, 5 Jun 2011, vip killa wrote:
http://pastebin.com/vxGM2n5j
We are getting those errors 100x per second in console when AGI set
debug is on
Can someone advise how to fix this? Thank you.
Don't request 'WAIT FOR DIGIT 1000' from a dead channel.
Don't ignore the error from 'WAIT
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like
On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change
at some point, mid-call? Or do you mean: Will Asterisk properly
react to such a re-INVITE and change codecs if asked to do so by
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
and
Will Asterisk properly react to such a re-INVITE and change codecs if asked
to do so by the dialog counterparty?
On Tue, May 3, 2011 at 12:56 PM, Alex Balashov
On 05/03/2011 01:16 PM, Gary Graves wrote:
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
I don't know of a way to do that. I suppose it might be possible if a
call were asynchronously transferred to a SIP peer that had different
codec
Hi
very thanks, that's work
bye
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14,
In that situation, I've had to do a pickup macro that kind of primes
the audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's channel
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
Don't know
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps (or
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so
On Monday 24 January 2011 04:09:31 Olivier wrote:
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote:
Or is there another work around to get ssh console colors using the
Debian * 1.6.0.28 LSB init script?
I also tried 'nocolor = no' in the [options] section of asterisk.conf
with no effect.
Try running asterisk using
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote:
On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com
wrote:
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es
wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural death),
or
the kernel runs out of
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS -
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death),
Actually, no. This is part of a migration, and those are mostly customers'
secondary lines (which for the most part, aren't even active). We get a
lot of these bad logins because the retry times on the ATAs are quite
short.
Asterisk really *shouldn't* leave zombies around for every bad login, but
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server is restarted (and the zombies die a natural
death), or the kernel runs
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural
Manmohan wrote:
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google but didnt get
any proper results.
I am facing one issue in Web-meetme on the expiry of any conference
that we create.
I am having Web-meetme 4.0.2
Hi Dan,
I am currently on Asterisk 1.6.2.14.
Thanks Regards
Manmohan Singh
On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote:
Manmohan wrote:
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google
Manmohan wrote:
I am currently on Asterisk 1.6.2.14.
Do you have schedule=yes in meetme.conf? I incorrectly
remembered/thought that all of the Realtime features were
controlled by that option, only a small number, such as
end times and CDR logging
On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin
Hi Dan,
In meetme.conf the schedule=yes was commented, after removing its working fine.
But one strange thing had started now. I started getting segmentation fault.
following are the errors which i see in it:
warning: difference appears to be caused by prelink, adjusting expectations
: [asterisk-users] Asterisk 1.6 (Web-meetme)
Hi Dan,
In meetme.conf the schedule=yes was commented, after removing its working fine.
But one strange thing had started now. I started getting segmentation fault.
following are the errors which i see in it:
warning: difference appears to be caused
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
--
No you can't
On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten =
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in
On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO o.calv...@gmail.com wrote:
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
10 nov 2010 kl
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote: Good idea Paul.
My debug
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum br...@woollum.com
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9
Has anyone started using Firefox 4 beta versions? We started today and find
that many of the GUI's attached to Asterisk respond differently and in many
cases not at all? We have found that details cannot be saves and that the
screens become very unstable. While we appreciate this is a beta
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused to negotiate T.38
[Sep 13 00:46:02] WARNING[3283]: app_fax.c:223
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused
On 09/14/2010 04:33 AM, Stanislav Korsei wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
Why install 0.0.5? Its ancient. the world has moved on.
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote:
Can you recommend any specific solution to this problem or way to install
app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in
Try removing the quotes in your n(true) priority.
Thanks,
--Warren Selby
On Aug 2, 2010, at 7:40 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before
El 29/06/10 15:28, Mark Deneen escribió:
We are experiencing intermittent DTMF problems here, with the
following setup:
ITSP - PIX - Asterisk (g729, RFC2833 for DTMF).
I am running Ubuntu server 10.04, but Asterisk is compiled by us and
not installed from the software repository.
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina mmol...@millenium.com.cowrote:
I've experienced a similar DTMF issue with recent asterisk 1.4 versions
(1.4.32, 1.4.33.1), I'm not sure about 1.6.2.X. What happens here is
that the DMTF activated features, like disconnect (default *) or blind
, 2010 16:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Of Aksel Celasun
Sent: Monday, June 28, 2010 16:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 30, 2010 11:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Actually, I should simply have tried. I did need
-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Here is my only question left about parkinglots in 1.6. How does the
parkinghints=yes parameter work?
I've tried using core show hints , but there are never any hints. Even
when a call is actually parked
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, June 30, 2010 13:38
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
In 1.4 you set up the lots you want to monitor
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 28. juni 2010 21:39
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users]
Michael wrote:
I am attempting to setup Asterisk to work with Gtalk.
I am using the following versions:
Slackware Linux 12.0
Asterisk 1.6.2.9
GNU TLS 2.8.6
Iksemel (svn v25)
OpenSSL 0.9.8o
It all compiles however about 10 seconds after starting Asterisk it crashes.
If there is any
Hello Platt,
Thank you for help.
I have tested and it works fine.
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Hello,
Thank you for your reply.
The first proposed solution has resolved the problem for a test in the local
network. Another test is planned today later with a client in the same NAT
and another in the public internet with a public static ip address.
Do you have any advice for that case?
Thank you for your reply.
The first proposed solution has resolved the problem for a test in the local
network. Another test is planned today later with a client in the same NAT
and another in the public internet with a public static ip address.
Do you have any advice for that case?
mosbah.abdelkader wrote:
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have
set up a VPN connection between 2 SIP clients and Asterisk using x-lite.
Just a guess, set canreinvite=no in the sip.conf for each of the end points
Doug
--
Ben Franklin quote:
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have set
up a VPN connection between 2 SIP clients and Asterisk using x-lite.
The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn
tunnel.
When attempting to make a call between the clients,
http://hostseries.com/asterisk-cdr-logging-in-mysql/
http://www.asterisk.net.au/tutorial/10/
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql
http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk
On Fri, Oct 30, 2009 at 11:35 AM, Joseph
On Friday 30 October 2009 01:05:26 Joseph wrote:
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
module show is showing cdr_addon_mysql.so
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file
Thanks Prince (good links) and Tilghman.
I'm using Gentoo installation of Asterisk-1.6.1.8-r1 that just showed up on
portage.
I've emerged(installed) asterisk-addons and this file usually creates necessary
drivers and copy cdr_mysql.conf file into /etc/asterisk (it worked in past
verions 1.2
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