ViciBox actually gives you the option of using the 2.2.1 release or
SVN/trunk versions ViciDial
Also, ViciBox is the officially supported ISO installer of the ViciDial
project.
But, both ViciBox and ViciDialNow are Linux ISO installers that will give
you a functional ViciDial system.
Thanks,
On Sun, Jul 25, 2010 at 10:20 PM, Janu Mukherjee janu.mu...@gmail.com wrote:
I installed asterisk server in my linux box. I configured a user 1000 using
xlite and registered with asterisk server in the same linux box. I
Where on the network is this box?
configured one more user 1001 in other
Hi,
Anybody out there running Adit600s?
I have in my care an Adit600 channel bank connected to an old (version
1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk
(1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail.
I have attempted to add the slowsequence = yes line
From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Sunday, July 25, 2010 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Kevin Keane
Subject: Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk?
Kevin Keane wrote:
I recently inherited a Vertical
On Sun, Jul 25, 2010 at 08:06:55PM +0100, Faris Raouf wrote:
I'm having some mysterious problems installing 1.6.2.10 on Centos 4.8
(totally up to date). I can't see anything on Google or the list regarding
this issue, which I find a bit odd considering 1.6.2.10 was released a few
days ago. I'm
Hi,
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
freetds-bin, but, when I run configure and then make menuconfig in section
Call Detail Recording - cdr_tds it's disabled. It only writes that
Depends on:
Hi,
I've managed to acquire a few Cisco handsets (7905, 7920) and would like to
use them with Asterisk.
Rather than simply switching to the SIP firmware I thought I'd use these
with chan_skinny - partly because this is Cisco's primary firmware and
therefore the phones might be more stable, and
Hi,
I installed asterisk server in my system running linux. I configured a user
1000 using xlite and registered with asterisk server in the same linux
system. I configured one more user 1001 in another linux machine and this
user also got registered with asterisk. But i am facing two issues here.
On Monday 26 Jul 2010, Andraž wrote:
Hi,
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
freetds-bin, but, when I run configure and then make menuconfig in section
Call Detail Recording - cdr_tds it's
On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote:
Hi,
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
freetds-bin, but, when I run configure and then make menuconfig in section
Call Detail
Hello,
I've been looking for this on voip-info and this list threads, and I am
guessing I am not looking right.
What I need is the way to capture (and write to DB) the information on who
'picked' or 'received' the incoming call.
Here is the example of .rb file that is called from
Hello,
I want to optimize my registrations and calls of peers to my asterisk
with the following options in sip.conf:
---///---
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture
We are having good results with
maxexp 120
minexp 90
defexp 100
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more experienced with these settings to help me to
optimize connections
On 07/25/2010 02:47 PM, Richard Kenner wrote:
At what stage will there be versions of the G.729 codec, res_cepstal,
skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if
people using that software could participate in the Beta.
We don't normally produce versions of our
!
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= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
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On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f
did you also hav qualify and qualifyfreq?
Thank you for reply,
On Mon, Jul 26, 2010 at 1:55 PM, Faisal Hanif fai...@vopium.com wrote:
We are having good results with
maxexp 120
minexp 90
defexp 100
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
Apologies if this has been asked before.
Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1?
Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the
source files for 1.4.34 over the top of the existing 1.4.24.1 files.
Obviously, I will need to keep my config files
I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or
web interface I can use to help do this. I am currently using Asterisk 1.4
Tony
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Hi All,
I have a quick question with regards the pbx_lua module.
Would the lua dialplan have direct access to the odbc configuration
within Asterisk, those odbc connections/dsn's that are defined in
res_odbc.conf/extconfig.conf/cdr.conf?
Thanks
Bruce
--
On Mon, Jul 26, 2010 at 8:15 AM, Tony LaMear tlam...@indyzoo.com wrote:
I need graph the utilization of my t1s. Does anyone know of a plug-in, code,
or web interface I can use to help do this. I am currently using Asterisk
1.4
http://oss.oetiker.ch/mrtg/
--
Paul Belanger | dCAP
Polybeacon |
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
Subject: [asterisk-users] 'dirty' upgrade of 1.4
Apologies if this has been asked before.
Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1?
Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding
It gives me headaches trying to use databases with Asterisk. That being
said, IMO the best answer to your query is to use the FORKCDR command so
that the call will be split into legs. When the operator answers the
call, that will be leg 1. When the call is transferred to the desired
party, that
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We are not using qualify for the peers which are not on static IP and
registering to server.
Regards,
Faisal Hanif
//
On 7/26/2010 5:06 PM, Catalin S. wrote:
did you also hav qualify and qualifyfreq?
Thank you for reply,
On Mon, Jul 26, 2010 at 1:55 PM, Faisal Haniffai...@vopium.com
You should be able to compile the new version, stop asterisk then make install.
If you do not do make samples then your conf files will be left alone. Once you
have done make install you can the start asterisk again.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On
appreciated!
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use cacti
Regards,
Faisal Hanif
/Think about the environment before printing this mail /P/ Tænk på
miljøet før du printer denne mail/
On 7/26/2010 5:15 PM, Tony LaMear wrote:
I need graph the utilization of my t1s. Does anyone know of a plug-in,
code, or web interface I can use to help
you can use all asterisk dial-plan functions and application in lua
plus additional complete lua features. so answer is yes.
Regards,
Faisal Hanif
On 7/26/2010 5:34 PM, Bruce McAlister wrote:
Hi All,
I have a quick question with regards the pbx_lua module.
Would the lua dialplan have
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression
Hi Danny,
I understand (and welcome) the separate src directories. This would
allow me to 'revert' should I feel the need (assuming I can just
re-compile over each one). I just need to know if I can re-compile over
the existing first.
Thanks for your reply.
-Original
On 07/26/2010 11:57 AM, Alexander Aksarin wrote:
On 20:59 Fri 23 Jul , Steve Underwood wrote:
That's just how your images look for me, so I guess your problem is
described here http://www.soft-switch.org/spandsp_faq/ar01s09.html
Steve
Big thanks for your help, Steve. I tried feh,
I don't have such a centos 4.8 system handy to test with.
What version of 'make' do you have?
make --version
rpm -q make
In any case, please submit a report to http://issues.asterisk.org/
Thanks Tzafrir.
GNU Make 3.80
Make-3.80-7.EL4
I'll submit a bug report. I just can't
Thanks for the quick response, however, how would I access an odbc dsn
from the pbx_lua dialplan that has been defined in res_odbc.conf or
related odbc structures? I've not come accross any documentation on that
feature yet.
Any tips/info/links would be appreciated.
On 26/07/10 14:33, Faisal
On 10-07-26 08:15 AM, Tony LaMear wrote:
I need graph the utilization of my t1s. Does anyone know of a plug-in,
code, or web interface I can use to help do this. I am currently using
Asterisk 1.4
I've been looking at the OpenNMS project recently.
http://www.opennms.org
Leif Madsen.
--
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
Regards,
Faisal Hanif
On 7/26/2010 7:02 PM, Bruce McAlister wrote:
Thanks for the quick response, however, how would I access an odbc dsn
from the
://www.asterisk.org/hello
asterisk-users mailing list
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Hello,
as I'm looking for a solution (with asterisk 1.6.2) , my
investigations leaded to :
- res_ais = libais corosync. (each node need to run corosync / aiexec)
- res_jabber = libjabber iksemel. (each node need to be connected on
an XMPP server)
I've been able to make some successful
On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote:
On 07/26/2010 11:57 AM, Alexander Aksarin wrote:
On 20:59 Fri 23 Jul , Steve Underwood wrote:
That's just how your images look for me, so I guess your problem is
described here
When you run make, it compiles the binaries in the src directory. Once
it is done compiling stop asterisk. Running make install will copy the
compiled binaries into their respective folders on your system. Then
just start asterisk. If you need to revert, stop asterisk, run make
install in the old
On 07/26/2010 10:55 PM, Tzafrir Cohen wrote:
On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote:
On 07/26/2010 11:57 AM, Alexander Aksarin wrote:
On 20:59 Fri 23 Jul , Steve Underwood wrote:
That's just how your images look for me, so I guess your problem is
described
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified func_odbc.conf you'd be able to do a
query like:
exten =
On 10-07-26 10:45 AM, Mathieu wrote:
Hello,
as I'm looking for a solution (with asterisk 1.6.2) , my
investigations leaded to :
- res_ais = libais corosync. (each node need to run corosync / aiexec)
- res_jabber = libjabber iksemel. (each node need to be connected on
an XMPP
On 10-07-26 04:03 AM, Jonathan Hunter wrote:
However, I've come across a couple of showstoppers and am not really
sure where to go from here. I've raised bugs for both of them (#17680,
#17692) and had no response so far - have I perhaps overestimated how
much chan_skinny is in use these days,
On 10-07-25 11:50 AM, Administrator TOOTAI wrote:
Le 25/07/2010 02:11, Norbert Zawodsky a écrit :
Hello again!
Hi
after it being relatively quiet her for the last weeks, my Astrerisk
server was the target of 3 of that nasty REGISTER attacks during the
last days.
[...]
Do like most of us
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Zarko Zivanovic wrote:
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
You need to change *ALL* the # to $
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor
The problem we are having with Asterisk is when we initiate a call via a
Zap line and it goes out on a Sip line. When it goes out via Sip we hear
no sound until the party we are calling answers the line. If the call
were to go out Sip-Sip or Zap-Zap it works perfectly fine. It is only
with the
Ahh ok, so I am only able to access the application/functions that are
available to the dialplan.
I was wondering if it would be possible to access the handle of the odbc
connection directly from the lua dialplan.
On 26/07/10 17:10, Leif Madsen wrote:
On 10-07-26 10:34 AM, Faisal Hanif wrote:
Hi!
Depending on the version of Asterisk you are running you can call a macro
or an agi as option to dial. These will be called when the line is
answered and you can find the channel name of who answered.
Do as he says, look at the M option to Dial.
Philipp
--
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you
On 26 July 2010 17:27, Zarko Zivanovic outlaw...@gmail.com wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
No success. Anybody please help!
-Original Message-
From:
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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The problem we are having with Asterisk is when we initiate a call via a
Zap line and it goes out on a Sip line. When it goes out via Sip we hear
no sound until the party we are calling answers the line.
Search for progress and/or progressinband.
--
On 26 July 2010 17:17, Leif Madsen leif.mad...@asteriskdocs.org wrote:
Unfortunately the developer who was looking after that channel driver
(community
developer) has been pulled off onto other projects it seems, so currently
there
isn't much support for chan_skinny.
If your timeframe is
I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything
looks fine, except the machine restarts the firewall whenever the DHCP lease
is renewed, thus flushing all the fail2ban rules (I think.). It seems to me
that a quick fix would be to have the system restart fail2ban
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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On 26/07/10 13:15, Tony LaMear wrote:
I need graph the utilization of my t1s. Does anyone know of a plug-in,
code, or web interface I can use to help do this. I am currently using
Asterisk 1.4
*Tony *
I've been looking at ZenOSS, which appears to have an asterisk zenpack
as well.
On 7/26/2010 1:40 PM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
This is what I do with Perl AGI scripts and it works fine. You
On Mon, 26 Jul 2010, Andres wrote:
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like: $message=/bin/echo my variables
On Mon, 26 Jul 2010, Zarko Zivanovic wrote:
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
I'm just a c weenie, but that syntax would execute a command named
$message, not
I use a custom script that I run using SNMP, and graph that using cacti.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce
McAlister
Sent: Monday, July 26, 2010 13:57
To: asterisk-users@lists.digium.com
Subject: Re:
On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrenga li...@torrenga.com wrote:
I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything
looks fine, except the machine restarts the firewall whenever the DHCP lease
is renewed, thus flushing all the fail2ban rules (I think…).
Randy R wrote:
On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com
wrote:
I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything
looks fine, except the machine restarts the firewall whenever the DHCP lease
is renewed, thus flushing all the
On Mon, 2010-07-26 at 12:10 -0400, Leif Madsen wrote:
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified
On Mon, 2010-07-26 at 17:43 +0100, Bruce McAlister wrote:
Ahh ok, so I am only able to access the application/functions that are
available to the dialplan.
I was wondering if it would be possible to access the handle of the odbc
connection directly from the lua dialplan.
Currently there
On Mon, Jul 26, 2010 at 12:19 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Why isn't the Asterisk box on a static IP on the LAN? That seems to be
asking for trouble using DHCP.
I was assuming he meant the ISP DHCP renewal.
/r
--
On Monday 26 July 2010 14:19:58 John Novack wrote:
Randy R wrote:
On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com
wrote:
I have tried to setup fail2ban on a machine running OpenSuSE 11.
Everything looks fine, except the machine restarts the firewall whenever
the
Hi guys,
i'm trying to use the featuremap of features.conf inside the app meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Subject: [asterisk-users] MeetMe
toca_macaco = 123, peer, Playback,tt-monkeys
But, if, inside the room, I press 123 the sound file tt-monkeys it's not
executed.
Danny,
didn't work... I didn't find other option to make meetme accpet dtmf but
F.
On Mon, Jul 26, 2010 at 5:25 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo
I think there is a mis-communication here; If you changed features.conf so
that toca_maccao = 123 . is now toca_maccao = 9, then if you press 9,
monkeys should play.
--
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Monday, July 26, 2010 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fail2ban -
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a
week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1
VPMADT032 Module, hooked up to 5 analog lines. I get the error message
referenced in the subject in my dmesg output everytime I load / reload DAHDI
Jonathan wrote:
I've managed to acquire a few Cisco handsets (7905, 7920)
and would like to use them with Asterisk.
Rather than simply switching to the SIP firmware I thought
I'd use these with chan_skinny - partly because this is
Cisco's primary firmware and therefore the phones might be
On 07/26/2010 05:48 PM, Warren Selby wrote:
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as
of a week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules
and 1 VPMADT032 Module, hooked up to 5 analog lines. I get the error
message referenced in the subject in
Manmohan Singh Jandu wrote:
Excellent!
I finally got it working, it was ODBC drivers issue
actually. Installed the proper compatible version and its working.
I thought that might be the case.
There are still few errors which i see on asterisk console:
[Jul 19 13:58:51] WARNING[30213]:
Manmohan Singh Jandu wrote:
OK, now i added the column members in the table booking manually.
and disabled selinux to have this working.
Now i am struggling with recording option in webmeetme.
Not sure on how to enable it, though m checking the checkbox
while creating the conference. But
That was exactly what I did...but didn't work if I insert the option
p in the meetme on the dialplan, I can leave the room pressing #, so dtmf
is working fine
On Mon, Jul 26, 2010 at 6:07 PM, Danny Nicholas da...@debsinc.com wrote:
I think there is a mis-communication here; If you
Hi Dan,
On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote:
I'll dig around in my archives to see if I can find my old patches
for either of these.
Many thanks - I'm happy to test patches if I can do so. At least I can
contribute in that way, even if I'm not directly contributing
Jonathan wrote:
On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote:
I'll dig around in my archives to see if I can find my old patches
for either of these.
Many thanks - I'm happy to test patches if I can do so. At least I
can contribute in that way, even if I'm not directly
I have asterisk running at home, a friend would be traveling out of the
country and I want him to be able to put a call through from his remote
location, I am wondering how I would configure the X-lite client on his pc so
he would be able to call through assuming my public address is A.B.C.D
Configuring x-lite is a smaller problem here, do you have on your router
your public IP ported to private IP at all and have you tested it before?
As for x-lite check it on my website at
http://visionvoip.com/help/x-lite.php
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-26 8:51 PM,
To have your asterisk box reachable from internet you must configure
static nat on your router to get sip traffic to the public Ip
redirected to your internal ip. Make sure that sip and rtp traffic are
not bloked by firewall.
And configure xlite to connect to your public ip address.
On Mon, Jul 26, 2010 at 6:14 PM, Adolphe Cher-aime achera...@gmail.com wrote:
To have your asterisk box reachable from internet you must configure static
nat on your router to get sip traffic to the public Ip redirected to your
internal ip. Make sure that sip and rtp traffic are not bloked by
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hi Guys,
I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.
The Asterisk Queue Analyzer is to serve as the graphic tool for call center
or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR
to create a graphic
You may need to add r as option perameter to dial command.
Regards,
Faisal Hanif
On 7/26/2010 9:39 PM, Chris Ramirez wrote:
The problem we are having with Asterisk is when we initiate a call via
a Zap line and it goes out on a Sip line. When it goes out via Sip we
hear no sound until the
Hi,
I have tried number of time if we update any CentOS system (or use
latest CentOS version) then compile asterisk 1.6.2 with pbx_lua support,
asterisk will crash on starting and will give a core dump.
Issue is easy to produce,
Install latest CentOS on a system.
Install LUA LUA Headers
Hi,
I have xlite registered with a user. Now i dial an extension say 1500 which
has the dial plan as follows.
exten==1500,1,AGI(localhost//hello.agi
So when i dial extenstion 1500 the script hello.agi is invoked which in turn
plays a welcome message. I now want to transfer the call now to
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