On Sun, Jul 25, 2010 at 3:11 AM, Norbert Zawodsky wrote:
> Hello again!
>
> 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. While I can see not much danger coming from these attacks (I
> use
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 oper
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 u
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 p
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 bar
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
On Mon, Jul 26, 2010 at 6:14 PM, Adolphe Cher-aime 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 firewall.
> And
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.
Adol
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, "ayode
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 a
Jonathan wrote:
>> On 26 July 2010 23:50, Dan Austin 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
> wo
Hi Dan,
On 26 July 2010 23:50, Dan Austin 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 wonderful code
modul
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 wrote:
> I think there is a mis-communication here; If you changed features
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. B
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[3021
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
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
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
-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 - SuSEfirewal
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api
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 wrote:
> *>From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo
> >*Subject:* [ast
>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.
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
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. Torrenga
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 lea
On Mon, Jul 26, 2010 at 12:19 PM, John Novack
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
--
_
-- Bandwidth
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 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 func
Randy R wrote:
> On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrenga
> 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
On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrenga 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…). It seems to me
>
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: [aste
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
$me
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 variabl
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
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.
http:
rt with ${ vs #{
>>
>> Unless that is some special indication in SQL that I'm unfamiliar with.
>>
>> Leif Madsen.
>>
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provide
vided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__
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 whenev
On 26 July 2010 17:17, Leif Madsen 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 just a week or so, you'
> 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".
--
___
ter
the call is over:
2) Call AGI after call is over
In case 1) you'll need the following somewhere in your script after
the call is answered or completed.
$agi.execute('Set(LOC=${CDR(dstchannel)})')
In case 2) just put
Set(LOC=${CDR(dstchannel)})
in your dialplan before cal
On 26 July 2010 17:27, 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}")
>
>
>
> No success. Anybody please help!
>
>
> -Original Message-
> From: asterisk-users-b
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 sug
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
--
__
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:
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
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 Libert
ttp://lists.digium.com/mailman/listinfo/asterisk-users
______ Information from ESET NOD32 Antivirus, version of virus signature
database 5314 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
______ Information from ESET NOD32 Antivirus, versi
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.
>>
> [...]
>
>
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 day
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 XM
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 => start,1,
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
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 s
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 http://www.soft-switch.org/spandsp_
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
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET NOD32 Antivirus, versi
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 pbx_l
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.
--
__
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
>
> 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
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 f
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 Mess
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
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 direc
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
7f into answeredby.
Any help is greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, v
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 Jul
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 Hanif wrote:
We are having
gards,
> Steve
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.or
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, tha
>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 rebuildin
On Mon, Jul 26, 2010 at 8:15 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
>
http://oss.oetiker.ch/mrtg/
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabbe
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
--
__
r update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET NOD32 Antivirus, version of virus signature
database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.
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 (a
did you also hav qualify and qualifyfreq?
Thank you for reply,
On Mon, Jul 26, 2010 at 1:55 PM, Faisal Hanif wrote:
> We are having good results with
> maxexp 120
> minexp 90
> defexp 100
>
> qualify = yes
> qualify = 500
> qualifyfreq=5
> registerattempts = 0
> registertimeout = 10
> maxexpiry
> 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/ope
_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id}")
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus signature
database
greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
--
_
-- Bandwidth and Co
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 bin
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 fro
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 t
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 experien
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 extension
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 R
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_td
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.
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 pa
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 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.
91 matches
Mail list logo