Dears,
1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes does this GSM
card, ASR ,ACD on each card.
It's too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks
How
Hi
Can you please send me a copy of the AGI script you wrote, in order to have
look on it, it seems this is a solution for my problem
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
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{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
Any update ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, June 21, 2011 12:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users
Dears,
i am using sipp to test asterisk(1.6.22) performance ,but when i limit the
calls to 150 ,only 100 active calls on asterisk found ?why
sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk call limitation
It could be your OS limit try ulimit command.
--
Sent from my iPhone
On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 06/20/2011 01:09 PM, Khaled W. Chehab wrote
and restart asterisk also you can set in limit.conf file
I had this issue before and I solved that way.
--
Sent from my iPhone
On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com
wrote:
I tried the ulimit
[root@localhost ~]# ulimit
Unlimited
Then
sipp -sn uac -d 1 -s 2005
(kbytes, -s) 10240
cpu time (seconds, -t) unlimited
max user processes (-u) 65536
virtual memory (kbytes, -v) unlimited
file locks (-x) unlimited
[root@localhost ~]#
-Original Message-
From: Khaled W. Chehab [mailto:kche
Dears
I already read most of post on asterisk group and
(http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning)
But I could not find a calculator
1-Is there a calculator I can download for that
2-What I the maximum simultaneous calls that can asterisk handle using CPU
3.0
perl libraries are so fast to manage/debug and easy to use,more over you can
call too many function from system, and its good documented .
Perl is the best J
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote
i installed skype for asterisk
i can send and recieve calls normaly
how can i receive messages from another skype user
i Succeed to send only
using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message
text)
how to receive messages using this code
= us...@gmail.com,1,Dial(SIP/102)
It doesn't matter the context in gtalk or jingle ,..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Friday, February 25, 2011 2:30 PM
To: 'Asterisk Users Mailing
-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message
Install asterisknow and begin from there.
http://www.asterisk.org/asterisknow/
and don’t miss to read the documentation
https://wiki.asterisk.org/wiki/display/AST/Home
Regards
Khaled Chehab
NGN Eng.
Operations Office - Lebanon
Office : +961 1 868686 ext 115
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Friday, December 17, 2010 6:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Attack problem
On Friday 17 Dec 2010, Khaled W. Chehab wrote:
HI,
My system been attacked from
Hi,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
Thanks ,it solved by adding
insecure=very
regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, September 28, 2010 2:16 PM
To: Asterisk; Asterisk List
Subject: [asterisk
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk Digium E1 R2
ProtocolàCisco E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
configuration ?
Hi,
how to write the cdr directly to the databse (Mysq)instead of importing
Master.csv to table using a php script.
Noting that I load asterisk_addons_mysql
rev-xx-xx-xx-xx*CLI cdr status
rev-xx-xx-xx-xx*CLI
Call Detail Record (CDR) settings
--
Hi,
I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between digium card E1 to test the
configuration of dahdi
What I want to do scenario is
I connect port 1 and port4 in the digium card with E1 cable
SIPcall--E1
Find my dahdi config files below
dahdi-channels.conf
; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4
ClockSource
group=0,11
context=default
switchtype = euroisdn
signalling = pri_cpe
channel = 1-15,17-31
context = default
group = 63
; Span 2: TE4/0/2 T4XXP
Hi,
I have a digium card (digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between spans on digium card in order to test
the spans.
I connect port 1 and port4 with cross E1 cable
I am trying to do this scenario
SIPcall-- Digium span
Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ?
And how to integrate
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail:
When we can expect to have a res_fax and res_fax_degium module for asterisk
V 1.6.2
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
I have problemin g729 codec compatibility,I get the g729 module from
http://asterisk.hosting.lv/ and I have Asterisk 1.4.22-3 RPM
What g729 module should I download ?
I already downloaded
codec_g723-ast14-icc-glibc-pentium4.so
[trixbox1.localdomain asterisk]# cat /proc/cpuinfo
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service ,
HI all ,
I am using ,Dial(SIP/Gateway/${EXTEN},m)
how can i modify asterisk, if it detects two early media to stop OR MUTE
the first RTP early media AND let the user hear the second early media
any one developed something like that or know from where I can do this from
chan_sip.c?
Dears I installed digium fax and followed the instruction at
http://downloads.digium.com/pub/telephony/fax/README,And as you can see
above that t38 is loaded
I am using a call file to send fax1.tif file as fax to the gateway named
add
The problem that Addpac send always Receive 488 Not
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service ,
Hi.
Does asterisk support muting per a specific channel?
(like the soft hangup command, were you specify a channel and then
asterisks hangs it up).
1-If it does not, how will one go about to do something like this?
2-how to let the user hear 183 the early media like voice mail prompt
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
Dears,
When my GW send a call to asterisk v 1.4.24 ,
Asterisk send Status: 420 bad extension (unsupported)
Why? Any modifications should be done one sip.conf
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
Dears
How to disallow asterisk to send the keep alive 200 ok message to the peers
and trunks.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing
-my idea is to add music on hold stop when asterisk detect --
SIP/OPNS-096456c0 is ringing line
In
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
Dears
-How can I stop MOH when status of the dial is ringing and let the user
hear
the Ring Back Tone from the termination Gateway.
Remove the 'm' out of your dial command:
m
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 8:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
Dear Ben,
I tried a lot ,Kindly can you give me an example
] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 9:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
Thanks for answering Doug
I am using exten = _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
Change
: [asterisk-users] MOH
Khaled W. Chehab wrote:
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
I don't think you'll be able to mix and match via the dial application.
You may have to try using AGI for this. That, I can't help you with.
Doug
message. At the same time the
code that skips passing the ringing to A-leg
has to be disabled.
Martin
On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab kche...@xplorium.com
wrote:
Dear Martin
Can you inform me how to make the patch or from where I can get it
otherwise
if there is an application
the 180 Ringing being passed if
you have 'm' option to Dial and you do.
Try to take the 'm' out and see if 180 Ringing is passed to the A-leg.
So if you want MOH and then when 180 Ringing comes to turn it off =
you need a patch.
Martin
2009/4/4 Khaled W. Chehab kche...@xplorium.com:
10x Martin
Dears
Asterisk is a median server between the caller and the terminations gateway
The caller send the call to asterisk à asterisk will play music on hold
untill the termination gateway send 200 OK and the RTP establish
My problem that, Asterisk is not forwarding the 180 ringing from the
you patch
it.
DISCLAIMER: I may be wrong and was wrong before.
Martin
On Thu, Apr 2, 2009 at 11:07 AM, Khaled W. Chehab kche...@xplorium.com
wrote:
Dears
Kindly find my dial script below,I am trying to send the caller 180
ringing
but all tries were failed,
The caller always receive 183
)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
Dears
How can I send or force sending 180 Ringing instead of 183 back to the caller
?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding
Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its
] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original
() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users
Dears,
- Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.
Regards
*
No employee or agent is authorized to conclude any binding
YMMV, but you might try this
Exten = s,1,background(background_song)
Exten = s,n,Answer() ;start billing
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
Dears
What's the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail:
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Differences
1.6.0.6
- 1.4.23
--
0.1.77.6 :-)
http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?revision=172635v
iew=co
klaus
Khaled W. Chehab schrieb:
Dears
What's
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