Dears,
My scenario is to accept the call from user àAnswer the call -àplay mohà
dial(SIP/Trunk,X)
The problem is when the user send the bye the trunk call will not hangup
How to solve this issue
exten => 446696,1,Ringing
exten => 446696,n,Answer()
exten => 446696,n,Wait(2)
ext
How can I do that with asterisk .
2-Do any one know from where I can download a2billing prompts in Arabic for
free.
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +
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
{\*\htmltag244 }
{\*\htmltag84 }\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 }\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 }
{\*\htmltag64 }\htmlrtf {\htmlrtf0
{\*\htmltag244 }
{\*\htmltag84 }\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 }\htmlrtf\par}\htmlrtf0
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: [aste
(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
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"
wrote:
>
> I tried the ulimit
>
> [root@localhost ~]# ulimit
> Unlimited
>
> Then
> sipp -sn
89] file.c: Failed to write frame
[Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame
[Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame
Khaled Chehab
NGN Eng.
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
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
MHZ and 4GB ram
With rtp g729 and there is no codec transcoding ,
3-And what is the number of simultaneous calls if I use direct RTP
(Canreinvite=no /Directrt=yes)
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
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
Can anyone make it more clear please
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Can you please send me a how to please or a simple lines?
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: <mailto:kche...@xplorium.com>
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=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten => s,n,NoOp(Received message: ${message})
any idea
regards
add=true
;buddy_presence=no
mohinterpret=default
;mohsuggest=none
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: <mailto:kche...@xplorium.com
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 http://s
Original Message-
From: asterisk-users-boun...@lists.digium.com
[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 De
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 Xplo
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 Xploriu
{\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 http://s
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 http://s
{\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 http://s
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
--
Log
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
CI) Card 0 Span 3RED 0 0 0 CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
T4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
Khaled Chehab
NGN Eng.
Untitled
Oper
4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS
HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: <mai
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: <mailto:bs...
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
proc
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 , t
HI all ,
I am using ,Dial(SIP/Gateway/${EXTEN},m)
how can i modify asterisk, if it detects two early media to stop OR MUTE
the first RTP early media AND let the user hear the second early media
any one developed something like that or know from where I can do this from
chan_sip.c?
regard
Dears I installed digium fax and followed the instruction at
http://downloads.digium.com/pub/telephony/fax/README,And as you can see
above that t38 is loaded
I am using a call file to send fax1.tif file as fax to the gateway named
add
The problem that Addpac send always Receive 488 Not acce
Hi
I use dial with music on hold command
exten => _X.,n,Dial(SIP/Trunk/${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 , t
Hi
I use dial with music on hold command
exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of servic
Hi.
Does asterisk support muting per a specific channel?
(like the soft hangup command, were you specify a channel and then
asterisks hangs it up).
1-If it does not, how will one go about to do something like this?
2-how to let the user hear 183 the early media like voice mail prompt since
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
Dears,
When my GW send a call to asterisk v 1.4.24 ,
Asterisk send Status: 420 bad extension (unsupported)
Why? Any modifications should be done one sip.conf
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
: [asterisk-users] MOH
Khaled W. Chehab wrote:
> Man :)
> I want the MOH play until Asterisk receives 180 ringing or 183 from the
> termination GW.
>
I don't think you'll be able to mix and match via the dial application.
You may have to try using AGI for this. That, I can&
] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 9:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
> Thanks for answering Doug
>
>
> I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macr
oun...@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 yo
: 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
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 whic
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 writte
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
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 can
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
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Sunday, April 05, 2009 5:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Hi Khaled,
app Dial clearly is coded to ignore the
Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Hi Khaled,
I believe the 180 Ringing will be sent only if your B-leg sends it to
Asterisk.
Asterisk doesn't know WHEN the call will physically ring the destination
number
so unless you GW tells it you won't ever see that mess
e "183 Session Progress" with "180 Ringing" (line 3950 in my source)
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, Apri
wer() 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:
um.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] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will start
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 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I t
e "183 Session Progress" with "180 Ringing" (line 3950 in my source)
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, Apri
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'
Subje
rcial 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: '
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 agree
ers] Early Media
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, Mar
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 a
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=172635&v
iew=co
klaus
Khaled W. Chehab schrieb:
> Dears
>
>
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: <mailto:
Dear All
FBI issues VoIP security warning on Asterisk -- but which version?
Any one know which version ?
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without expr
Dears
I have TE405P digium card connected to my asterisk pbx.
Kindly can you assist me my a sample from where and how I can configure and
limit the incoming calls to E1 channels in the direction of to be able to
route the incoming calls.
Regards
erisk is a good
way to start Asterisk.
Thanks,
Steve Totaro
On Fri, Nov 14, 2008 at 8:57 AM, Khaled Chehab <[EMAIL PROTECTED]> wrote:
Dear All
I tried to stop asterisk and start it with debugging ,kindly heck the
results
Already I modified asterisk.conf
astrundir => /var/run/aste
f Of Godson Gera
Sent: Friday, November 14, 2008 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk/E1
On Fri, Nov 14, 2008 at 6:29 PM, Khaled Chehab <[EMAIL PROTECTED]> wrote:
Dear All
I installed a Digium card TE405P with za
Dear All
I installed a Digium card TE405P with zaptel and its running successfully
with no alarms, but asterisk is not running .
Any one have a cure or advice
03:09.0 Communication controller: Digium, Inc. Wildcard TE405P quad-span
T1/E1/J1 card 5.0V (rev 02)
Nov 14 07:56:58 localhost ke
I want to have a PC-based real-time VoIP bulk call generator (including both
SIP signaling and RTP generation)
for stress testing and precise analysis of the VoIP network equipment.
Do any one knows a free program can do that .
Regards
***
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No
215.96.142.83(None) caac0846-cf 00101/0 unkn No
192.168.8.106(None) 94910146-46 00101/0 unkn No
192.
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No
215.96.142.83(None) caac0846-cf 00101/0 unkn No
192.168.8.106(None) 94910146-46 00101/0 unkn
Hi All
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail w
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without
any one know a distribution contain asterisk have solution like that ?
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab
Sent: Tuesday, January 29, 2008 10:36 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discu
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 . and forward it to a media gateway ..
Regards
Khaled chehab
*
No
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 . and forward it to a media gateway ..
Regards
Khaled chehab
*
No employee or
Dears
Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on
centos with kernel 2.6.9-55.EL.
Can you please send me the documentation link on how to or write down how to
.
Regards
*
No employee or agent is aut
Dears
Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on
centos with kernel 2.6.9-55.EL.
Can you please send me the documentation link on how to or write down how to
.
Regards
*
No employee or agent is author
I am using asterisk 1.4.5 with asterisk-addons-1.4.2
On /var/log/asterisk/cdr-csv/Master.csvthe unique id showed
But in mysql database ,the unique id is not shown ,how can I fix it ..
Regards
_
*
No employee or agent i
I am using asterisk 1.4.5 with asterisk-addons-1.4.2
On /var/log/asterisk/cdr-csv/Master.csvthe unique id
But in mysql database ,the unique id is not shown ,how can I fix it ..
Regards
*
No employee or agent is authorized to concl
-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
Do you have MySQL installed in your machine???
On 6/21/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:
>
>
>
>
> No one faced a problem like this !!
>
>
>
> __
No one faced a problem like this !!
_
From: Khaled Chehab [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 21, 2007 12:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Subject: asterisk 1.4.1 app_addon_sql_mysql
I am using
I am using centos 4.4 updated using yum
when I enter asterisk-addons-1.4.1 directory and make menuselect
*
Asterisk-addons Module
Selection
*
when I enter asterisk-addons-1.4.1 and make menuselect
*
Asterisk-addons Module
Selection
*
Press 'h' for help
I have question concerns asterisk
1-What is difference between G.729 and G.729A?
2-How can I know the requirement hardware for 150 extension on asterisk
1.4.4 making 50 simultaneous call?
3-Do asterisk have a codec conversion?
Regards
*
Any one knows how to make Meet Me video conferencing room.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an o
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
... error appears >
>From where I can get the missing rpms .or kernel source
>From where I can get the centos 4.4 server kernel source.
Regards
*
No employee or a
- Non-Commercial Discussion
Subject: Re: [asterisk-users] yum om centos
independently install each rpm via rpm command :-/
On 04/06/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I have 2 servers, one connected to internet and the other is on a private
> lan have no access
server I tried to update using
yum update
but the yum update failed.
How can I do that with out connecting the second server to internet .
Khaled
Regards
*
No employee or agent is authorized to conclude any binding
I already did what you said,please see the results when you compile using
make
ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type
/usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencing pointer to
incomplete type
/usr/src/zaptel-1.4.1/pciradio.c:1535: error: dere
I already did what you said,please see the log results in zaptel.rar
attached when I compile zapltel using
make
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without exp
I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there
is someone did that, Please in need to have the installation procedure step
by step. Its too urgent for me .
Thanks alot
Regards
*
No employee or agent is authorized t
I am trying to make a mirroring for my asterisk using nextone SBC,I have a
problem ,which is when and end point send Invitation to SBC realm .
This realm is send INV and REG messages to Asterisk. Asterisk sends INV
message again to this realm.
NexTone SBC try to send again to asterisk and
I am trying to make a mirroring for my asterisk using nextone SBC,I have a
problem ,which is when and end point send Invitation to SBC realm .
This realm is send INV and REG messages to Asterisk. Asterisk sends INV
message again to this realm.
NexTone SBC try to send again to asterisk and
I am using centos 4.4 ,when I am compiling zapltel using l make linux26
,error accrued ,what s missing
[EMAIL PROTECTED] zaptel]# make linux26
grep: /include/linux/autoconf.h: No such file or directory
make: *** No rule to make target `linux26'. Stop.
Regards
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
,the following error appears
>From where I can get the missing rpms .or kernel source ,or the kernel
header
Generating input for menuselect ...
grep: /include/linux/autoconf.h: No such file or directory
make[1]: Entering
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
,the following error appears
>From where I can get the missing rpms .or kernel source
grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file
or directory
grep: /lib/modules/2.6.18-8.el5/build/
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is
conf => 222| at meetme.conf
at meet_me_additional
like this
exten => 21,1,MeetMe(21,dq)
exten => 21,2,Playback(beep)
or this
exten =>
I have asterisk 1.4 ,the function that I am using in extensions.conf is not
functioning
Its was functioning on asterisk 1.2.further more
cdr_addon_mysql.so cdr_csv.so cdr_custom.so cdr_manager.so
cdr are loaded
Is there any missing module ?
Function IS
Any one knows where to install chan_sccp for asterisk 1.4 ???.
Please guide me from where can I download the asterisk 1.4 sccp channel
driver and how to install it because I tried to get
chan_sccp-mayday05.tar.gz
When I trying to install it ,error happened like this.
Please help me how to s
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