,
Faisal Hanif
--
_
-- 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.org/hello
asterisk-users mailing list
Hi,
I am also wonder that same SRV record is working fine on one machine but not
on 2nd while both have same asterisk version.
It may be some missing OS utilities which asterisk using to resolve SRV?
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun
You need to copy or soft link a2billing.conf to /etc/ folder as by default
latest version search for it in /etc/
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent
Try setting insecure=port,invite in sip peer config.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 15, 2010 9:14 PM
To: Asterisk Users Mailing List
Till now I am not able to find any difference between both machines.
Can you please tell me how I can try to resolve it on OS level using some
utility like dig?
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Both have CentOS 5.2.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Tuesday, June 15, 2010 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
If you use curl realtime for registrations you can add useragnet check
in your CGI and also lot of else as well.
Regards,
*Faisal Hanif
*On 6/29/2010 4:48 PM, Tarek Sawah wrote:
well there are two restrictions.. the IP address of the station they
are using it .. and the UserAgent..
one
Simply set it to costume field of cdrs in dialplan and you will have
it a part of native cdr
Regards,
*Faisal Hanif*
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Hi,
I am in process of merging all my AGIs+Dialplan to a single LUA
dialplan. It seems much interesting to me spacial LUA tables which allow
me to support a complete object like programming. Yet I did not
completed / tested.
Regards,
*Faisal Hanif*On 7/1/2010 2:37 PM, Gilles wrote
and
asterisk will use configuration returned by that external application
and will treat it same as in static file. Here you again have full power
of programming language in you hand.
Regards,
Faisal Hanif
On 7/7/2010 1:08 PM, Hans Witvliet wrote:
On Wed, 2010-07-07 at 12:12 +0600, ABBAS
,
Faisal
Hanif
VoIP Manager
m
+45 72 72 00 01
m
+92 32 1405 9996
Vopium A/S
| Office
Do some R D with asterisk function AMD (Answering Machine Detection)
if that can help you.
Regards,
Faisal Hanif
On 7/9/2010 11:24 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
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
You need to do it by manager interface
Regards,
Faisal Hanif
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
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
there's a variable
capturing who answered.
Zarko
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Faisal
Hanif
*Sent:* Monday, July 26, 2010 12:57 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] URgent
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
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
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
using YUM.
Download and Compile latest release of asterisk 1.6.2.
Try to start start asterisk in console mode.
It will crash on LUA and will give a core dump
Did any one got it solved? If yes how?
Regards,
Faisal Hanif
Hi,
It is simple to use max_limit perameter in dial command.
Regards,
Faisal Hanif
On 8/9/2010 2:01 PM, Catalin S. wrote:
Hello,
I wish to make a simple system to limit peers at x minutes depending
of buyer voip packet. Can someone help me with some directions?
I intend to make a separate
lback file to sent alert
call.
Signatures fai...@vopium.com
Regards,
Faisal
Hanif
VoIP Manager
n 8/9/2010 11:56 PM, Felipe Figueiredo wrote:
Hi guys,
is t
Hi,
SER is a most powerful SIP router but a SIPp is a VoIP load generation
software. So both are totally different and can not be used interchangably.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S//
On 8/10/2010 10:44 AM, kamrun nahar bina wrote:
Dear all,
What is the difference
to test any VoIP
platform.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S //
On 8/10/2010 11:33 AM, kamrun nahar bina wrote:
Dear Faisal Hanif,
Thanks for your reply.
What is the purpose of using SER ?
What is the purpose of using SIPp -I know little bit about this.
But I know nothing
Hi,
We are using 4-PRI card from http://atcom.cn for our development LAB and
we are satisfied with performance. It is also cheaper then other
products. They also have analog.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S
On 8/10/2010 6:40 PM, Jeremy Betts wrote:
I have always had very
read the value of var ${HANGUPCAUSE} next line to dial command.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S
On 8/10/2010 9:51 PM, bruce bruce wrote:
Hi Everyone
Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to
Bell Canada.
User claims that call hangup without
If Caller party hangups next to dial line wil not be executed but
control will hit to h extension of fame context but if Called party
hangups next to dial ine will be executed.
Faisal Hanif
On 8/11/2010 10:16 AM, Philipp von Klitzing wrote:
read the value of var ${HANGUPCAUSE} next line
(${WHOHAVEHANGED} have hanged the call reason is
${HANGUPCAUSE})
Regards,
Faisal Hanif
On 8/12/2010 12:29 AM, bruce bruce wrote:
Sorry, I am not following:
*//**/read the value of var ${HANGUPCAUSE} next line to dial command./*
*/
/*
*/Where is that value? Next to dial you mean right when
did you copied rc.redhat.asterisk script from contrib/init.d/ forlder
to /etc/init.d/ folder?
Regards,
Faisal Hanif
On 8/16/2010 2:28 PM, unsero...@aol.com wrote:
No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or
start the deamon
the way it works with 1.6.1.20
,
Faisal Hanif
On 9/11/2010 4:47 PM, Olivier wrote:
2010/9/10 Hose hose+aster...@bluemaggottowel.com
mailto:hose%2baster...@bluemaggottowel.com
Does anyone have a suggestion on how to handle this? For example,
if I
have a list of numbers that I want to go out a certain sip channel
Allow anonymous SIP and enable debug then check if calls coming from
same IP which you have configured in peer?
Regards,
Faisal Hanif//
On 9/11/2010 8:07 AM, bruce bruce wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working
use CACTI
On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do so but the average is done
on time or day ...
I use Mysql CDR
Thank you,
Mickael
--
for me up to a load 100 calls. It may work more but I haven't tested it.
Asterisk and Billing-Server was running on separate machines.
For further help you can call me (as you know my number :P).
Regards,
Faisal Hanif
Hi,
I have used 4-PRI card from atcom.cn and it works perfectly for me.
Regards,
Faisal
+923214059996
On 12/27/2010 12:25 PM, Asim Amin wrote:
Hello All,
Anyone who has experience using Digium analog card clones from any of
the following:
1. Zycoo
2. CTVON
3. Chinaroby
4. Etross
5.
Hi,
1-TuneUp your setting in /etc/dafult/asterisk
2-Stop l;oadng all not required modules by adding noload =
modulename.so lines to /etc/asterisk/modules.conf
Regards,
Faisal
On 12/31/2010 7:59 AM, Larry Wimble wrote:
Asterisk gurus
I just installed asterisk 1.8.1.1 along with
The settings you are asking varies in different countries and providers. You
need to contact you provider for it.
From: Roi Stork
Sent: Thursday, February 10, 2011 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] zaptel/dahdi settings for singtel E1
There are some flags in general settings of dialplan which enable/disable
modify this behaviors of dialplan. Have a look on sample extensions.conf for
general tab settings. I will see if I can have time today to tell you exact
parameter name.
From: Dovid Bender
Sent: Thursday, February 10,
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
and use proper parameters to dial command to pass early media.
-Original Message-
From: Benoit Panizzon
Sent: Thursday, February 10, 2011 4:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Early audio SIP
Well. I suggest to use DB function instead of modifying asterisk source. You
can add one additional column and write and after-insert trigger in your cdrs
table which convert dattime to your required format and update the value of
added column.
From: Rodrigo Lang
Sent: Thursday, February 10,
Well I think you need major changes as application in android run in sandbox
instead of direct Linux APIs. Till now no news on it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Monday, February 14, 2011 6:46 PM
To:
Better to report a BUG to cisco.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller
Sent: Monday, February 14, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22
You may need to provide some more scenario detail
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, February 14, 2011 7:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] issue with
In case of asterisk you simply can't accept registration from an IP which
you have mentioned as static host for IP authentication.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:37 PM
You need to use relay request in your SBC instead of forward.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing
You may need to share your LUA code and the extension your call is need to
execute.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Pires
Sent: Wednesday, February 16, 2011 3:29 AM
To: Asterisk Users
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:
Check if dtmfmode
-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:
Check if dtmfmode is properly
. Can pluged to asterisk PBX
machine and used as FXO device.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan
Sent: Wednesday, February 16, 2011 10:49 AM
To: asterisk-users@lists.digium.com
Subject
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To:
It is in client but not in asterisk sip channel
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to diable echo
Did you executed Answer() before it?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] function Echo() doesn't work
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] function Echo() doesn't work
Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS
Stick). Just only no echo on SIP. Any suggestion?
2011/2/16 Faisal Hanif fai...@vopium.com
Did you executed Answer() before
-, ) in new stack
== Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on
'SIP/sipgate-account-'
here is the log. It is as same as I got from CAPI and Datacard. I just didn't
hear the echo from SIP connection.
2011/2/16 Faisal Hanif fai...@vopium.com
Check
in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] function Echo() doesn't work
I tried to set allow=all in sip.conf. But it still doesn't work.
2011/2/16 Faisal Hanif fai...@vopium.com
I faced same issue for sipgate but got it resolved by allowing all
You can do it using callback files or AMI.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu
Sent: Wednesday, February 16, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play one audio file to the
seconds.
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Carvalho
Sent: Wednesday, February 16, 2011 9:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] trunk not working
: Faisal Hanif
Subject: Re: [asterisk-users] trunk not working if I register a phone at the
same IP as the trunk peer's IP
Isn't this a limitation that can be surpassed with some configuration that
I'm lacking in my sip.conf or extensions.conf of my asterisk?
Ricardo.
On Wed
EAGI could be your target application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, February 16, 2011 11:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pipe audio
You can simply use Portable LinuxLive USB Creator 2.6 or grub4dos. And
make your USB bootable by any Linux Live ISO.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan
Sent: Wednesday, February 16, 2011 11:24 PM
To: Asterisk Users
Well you simple use dtmfmode=info in peer configuration of Snome phone.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 18, 2011 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
The difference you will feel when using callback files or AMI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, February 18, 2011 1:31 PM
To: asterisk-users@lists.digium.com
Subject:
This is not Digium's customer support address but free public emailing list
for asterisk user's contributed by community volunteers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Zieher
Sent: Friday, February 18, 2011 2:19 PM
Did you checked if you extension.ael doesn't have syntax error?
Did you upgraded anything after last compile?
Or
Try a clean recompile
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
From: asterisk-users-boun
: ast_compile_ael2
On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote:
Did you checked if you extension.ael doesn't have syntax error?
I think there is no error. I loaded the standard ael first (provided by
asterisk) then my test config, got the same result.
Did you
If you are using asterisk 1.8.x you don't need to type \ for spaces you can
write simple query and use spaces as normal it will work fine.
Faisal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: Friday,
If your PRI provider permit you to associate any ANI to any Circuit-ID you
can do this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, February 24, 2011 12:17 PM
To: Asterisk Users Mailing List -
You don't need to put quotes around AGI name.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don't remember exact name but there are two authorities which provide
real-time portability information online but you need subscription.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
www.numberingplans.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where
You can find lots by googling but none can give realtime stats as it depends
on network. Packet drop, retransmit, codec type will make lot of vibrations
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, March
Well a solution for you to put original context name in variable and then
use that variable in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Friday, March 04, 2011
1-Check signaling type on gateway PSTN ports
2-Set RTP timeout in SIP trunk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, March 04, 2011 7:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
AstPP jbilling
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, March 05, 2011 10:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
If you dialout call without answering and allow all codec for both peers
then codec negotiation will be direct between endpoints and asterisk will
only do media pass-through.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Sunday, March 06, 2011 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
This settings are for ISDN configurations I think.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Monday, March 07, 2011 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
AMI is single threaded link so waiting on it will bring things to hang mode
but FastAGI dialplan is multithread. Better to manage all info by AMI in a
local hash or array and use sleep/waiting on AGI till required info
populated to hash/array by AMI.
-Original Message-
From:
When you compile asterisk you can select multiple language files by using
make menuselect additionally you find lot of free sources on internet for
language files. Simply create a folder with language short-code in sounds
and then set channel's language variable to that short-code.
-Original
-users] [1.4] Reading phone number the French way?
On Tue, 8 Mar 2011 17:31:26 +0500, Faisal Hanif fai...@vopium.com
wrote:
When you compile asterisk you can select multiple language files by
using make menuselect additionally you find lot of free sources on
internet for language files. Simply create
It just have ACL concept. You can add permitted IPs List to any peer then
only from that IPs user can register. If you want to permit all you can add
0.0.0.0 to ACL
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR
Sent: Thursday,
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied
because of contact ACL
On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote:
It just have ACL concept. You can add permitted IPs List to any peer
, you have permit=172.16.16.0/24 whereas suggestion was
permit=0.0.0.0/0.0.0.0
On 3/10/2011 1:48 AM, RR wrote:
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote:
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use
Asterisk doesn't have all features of SBC like relay and forward request on
packet level but all depends on your scenario what you need.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, March 10, 2011
Try by reversing the line number of permit deny
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, March 10, 2011 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Fail2ban
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, June 16, 2011 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to secure our Asterisk server
It depend on Hypervisor. if it is full virtualization then it will not be
more than a part sharing from system resources depends on VM configuration
and processing load.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Asterisk-SNMP could be an option for u.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Friday, June 24, 2011 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor
If you can explain a bit more what exactly you need?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, June 27, 2011 9:16 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Have you installed sample configuration files package?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele
Sent: Monday, June 27, 2011 4:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] not find files
Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.
-Original Message-
Have you tried SIP session timer values in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject:
Hi,
I don't think there is a way for it inside asterisk but you achieve it by
adding static route in Linux routing table and make interface having that IP
as default route for the interested IPs traffic.
Regards,
Faisal
From: asterisk-users-boun...@lists.digium.com
When you make a call asterisk always create a channel named as below,
CheannelType/PeerName-uniquecode
Like
SIP/jon-312abf
So here jon is the peer name. This can help you to identify a peer as long
as A-Leg is active.
Regards,
Faisal
-Original Message-
From:
The problem you are reporting is not related to realm but can be context or
domain.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Tuesday, July 05, 2011 11:59 AM
To: Asterisk Users Mailing
as time on each
trunk can be monitored via any queue monitoring tool. !!
or better use queue_log in realtime DB
As per my view this is most easy and optimized approach while keeping all
possible data in queue logs. Hope this will helpful for you.
Regards,
Faisal Hanif
-Original Message
If the problem always related to some specific module then try clean
recompiling asterisk if it is with random modules then check you system RAM.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina
Berretta
Sent: Wednesday, July
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