Hello All,
I do have asterisk installed for a call centre with aheeva application and
i would like to know how to configure the sound for the inbound calls and if
there is any possibility for agent to receive a file with the phone number
and name of clients: For your information there is no probl
Hello All
do you have ant software in order to change the format from mp3 or wav to
gsm in order to using it in asterisk file
thank you so much for your help and support
Best Regards,
salah
--
_
-- Bandwidth and Colocation Pr
sion'
> *Subject:* Re: [asterisk-users] convert from wav or mp3 to gsm
>
> AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files.
> You can use LAME and SOX to change files between these formats.
>
>
> --
>
> *From:* asteri
Hello All
i need your help i have asterisk installed in my server (unix centos) and i
want to create php code in order to start the asterisk service
using a agent poste (windows xp ) without acces to the server with putty
thanks and regards
salah
--
___
thank you for your response but i need this solution as soon as possible
to restart the asterisk service via a agent poste
if ther is any solution please
thanks
2010/4/2 Tzafrir Cohen
> On Fri, Apr 02, 2010 at 03:41:57PM +0000, salaheddine elharit wrote:
> > Hello All
> >
n Fri, 2 Apr 2010, salaheddine elharit wrote:
> >>
> >>> thank you for your response but i need this solution as soon as
> possible
> >>>
> >>> to restart the asterisk service via a agent poste
> >>>
> >>> if ther is any solution please
i have installed asterisk and i have 3 sip 104 ,105 and 106
Now I can make the calls with theses sip without issue
I want to configure the outbound calls for these sips like that:
104 permission to call any number, but for 105 and 106 I want to specify
some numbers to call
Any help pleas
thanks for reply i have a dial plan but can you please give me an exemple
regrads
2011/5/27 mahesh katta
> you need to make dial plan .
>
> On Fri, May 27, 2011 at 2:59 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> i have installed asteris
>> Alex Balashov - Principal
>>> Evariste Systems LLC
>>> 260 Peachtree Street NW
>>> Suite 2200
>>> Atlanta, GA 30303
>>> Tel: +1-678-954-0670
>>> Fax: +1-404-961-1892
>>> Web: <http://www.evaristesys.com/>http://www.evaristesys.com/
&
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten => _0678922645.,1,Set(CALLERID(number)=520460587)
exten => _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta
> Remove the _ in front of your dialplan,like
> exten => 0678922645,1,--
>
> On Mon, May 30, 2011 at 11:00 PM, sala
OK thanks a lot for your help now all is ok :)
2011/5/31 salaheddine elharit
> Hello
>
> after remove the _ and put the number like that 0678922645,1, the issue has
> been solved
>
> thank you so much :)
> 2011/5/31 mahesh katta
>
>> Remove the _ in front
Hello,
We have 1 server installed with centos and asterisk and the sip in
configured we can do the external and internal call without issue
Now I have installed the same centos and asterisk 1.4 in second computer in
the same network.
My question if there is any possibility to create a trun
hello list
i have 1 server installed with asterisk centos and digium card
i have installed the same configuration in another unit but in this unit
there is no card installed
i have created a sip trunk between the 2 servers like that
in the server 1 with card
sip.conf
[asterisk1]
type=freind
h
hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
i can record all the calls inbound and outbound without problem.
but when i receive an inbound call from customer in IAX(1000) and i want to
transfer the call to other phone SIP(
l(SIP/223)
and in extensions.conf i have
exten => 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
exten => 223,n,Hangup();
thanks and regards
2011/6/16 Leif Madsen
> On 16/06/
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen
> On 16/06/11 09:20 AM, salaheddine elharit wrote:
>
>> thanks for your response
>>
>> the call
=> 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten => 223,n,MixMonitor(blah.wav)
> exten => 223,n,Dial(SIP/223)
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Thursday, Ju
hello list
i need to create a call files in order to do a click to call with
asterisk1.4
i want to use sip 223 in order to call phone number
i have created a file.call in var/spool/asterisk/tmp and i move it to
var/spool/asterisk/outgoing
but there is no call
please tell me if there i
hello liste
i have create an menu like below
exten => my_number,1,Ringing()
exten => my_number,2,Wait(4)
exten => my_number,3,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,2,Background(${sounds_path}welcome)
exten => #,1,Goto(menu,s,1)
exten =
thanks a lot now it's work correctly :)
2011/6/20 Warren Selby
> On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> [home]
>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>> exte
unds_path}relation-support)
exten => s,5,Queue(default)
exten => t,1,Hangup()
2011/6/20 Warren Selby
> On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> [home]
>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/ast
ou want your system to behave and i'll be glad
> to help.
>
>
> On Tue, Jun 21, 2011 at 9:22 AM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> Hello
>>
>>
>>
>> I have created the menu below, with this menu when I call 520460XXX
Hello list
in the below menu all is ok ,but by intermittence when i call my number 529
i hear the welcome message and i still hear this message even i click in the
# in order to go to the call [menu] he still in the welcome message
and also by intermittence i can hear the welcome message and
hi
you can use MixMonitor like below
in my case i use MixMonitor with SIP and i can record all calls inbound and
outbound
Best Regards
exten => 57,1,Answer()
exten => 57,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten => 57,n,Dial(SIP/223)
exten => 57,n,Hangup()
2011/6/30 Ishfaq Malik
>
Hello
With x-lite version 3.0 I can limited the time of call in
Option===>advanced===>network RTCP has been inactive for 30sec (by default)
with Eyebeam 1.5.19 i do the same but the call has not been hang-up
If there is any way to do that with Eyebeam or in extensions.conf
Thanks and r
Hi
i want to use timeout with asterisk 1.4 in order to hangup the outbound
calls after 25 sec
i call my mobile number 067xxx from my sip acount 223 and i want to
hangu up the call automatic after 25 sec but there is no hangup after 25
could you please help me
exten => 223,1,Set(TIMEOUT(a
what can i do in order to fix this issue
regards
2011/7/8 A J Stiles
> On Friday 08 Jul 2011, salaheddine elharit wrote:
> > i want to use timeout with asterisk 1.4 in order to hangup the outbound
> > calls after 25 sec
> >
> > i call my mobile number 067xxx fr
.@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > salaheddine elharit
> > Sent: Friday, July 08, 2011 6:43 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] timeout with outbound calls
> >
&
ew stack
== Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-6ec45a88'
== End MixMonitor Recording SIP/223-6ec45a88
srvradio*CLI>
2011/7/8 Eric Wieling
>
> Show us the CLI output of the failed call.
>
> > -Original Message-
> > From: a
hello list
My question if there is any way to display a name to the customer when i
call from my Number using asterisk 1.4
Exp: I call from my number 0520 and I want when the customer recive I
call from 0520 he sees in his phone a name “test”
Thanks and regards
--
_
hello
i want to know how to do in order to block all numbers bgin by 00 and all
numebrs begin by 1
i use sip account
thanks and regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
whatever extension.
>
>
> ---
> Marcelo Ellmann
> Freeddom Tecnologia e Serviços S/A
> +55 11 52133200 Ramal 1016
>
>
>
>
> - Original Message -
> From: "salaheddine elharit"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
09 PM
> Subject: Re: [asterisk-users] block all numbers begin by 00 and 1
>
> On Tuesday 09 Aug 2011, salaheddine elharit wrote:
> > hello
> >
> > i want to know how to do in order to block all numbers bgin by 00 and all
> > numebrs begin by 1
> >
> > i
in my case i use snom 320 and 370
i flow this link and i can do the pichup with any issue
http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Pickup
Regards.
2011/8/11 Ishfaq Malik
> On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote:
> > Ah, now this is interesting as one of our clients
Hello List,
i have a small issue regarding to some numbers when i call these numbers
using asterisk i got all times answer_machine ,but when i call these
numbers from my phone or any numbers i get the customer and i can speak
without any issue. The issue just when i call these numbers using my
dial(dahdi/1/5551212)
>
> After
>
> Exten => s,1,dial(dahdi/1/w5551212)
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Monday, August 15, 2011 1
any help please
2011/8/16 salaheddine elharit
> hello danny,
>
> i don't have dahdi iuse zap
>
> like below
>
> exten => 0661760924,1,Set(CALLERID(number)=520460587)
> exten => 0661760924,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten =&g
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Wednesday, August 17, 2011 5:44 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial
Hello list
i want to use pickup with sip and astersik 1.4
i configured all the inbound calls in 1 sip phone 224 and want to pickup the
calls using 222 SIP
Could you please see the code below and tell me what is wrong
NB when i make *8+ok i can pickup the call but i want to specify the n
ok thanks for you response
how can i do in order to fix this issue
regards
2011/9/6 Daniel Tryba
> On Tue, Sep 06, 2011 at 04:43:39PM +0000, salaheddine elharit wrote:
>
> [asterisk 1.4]
> > [agents]
> > exten => _2XX,1,Dial(SIP/${EXTEN})
> > exten =>
Hi
try to add this
exten => _90Z,1,Set(CALLERPRES()=allowed)
exten => _90Z,n,Set(CALLERID(num)=5631040)
if you get the same result that is means that you don't have permission from
your provider
kind regards
2011/9/26 C F
> Confirm with your provider that allow you to set ca
hello list,
i have one question if there is any way to know the model of the card diguim
installed
without open the server :)
thanks and Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Ne
..@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Monday, September 26, 2011 1:16 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] model of diguim card
>
> ** **
>
> hello list,
>
>
>
&g
> Give zttool a try...
>
>
> On Mon, Sep 26, 2011 at 2:32 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> thanks for your response but there is no dahdi becouse i have asterisk
>> 1.4 installed (zaptel)
>>
>> any help pleas
there is no dahdi i have asterisk 1.4 there is a zaptel instaed dahdi
when i put dahdi show status i get command not found
regards
2011/9/27 Anton Kvashenkin
> dahdi show status?
>
>
> 2011/9/27 salaheddine elharit
>
>> thank you evryone for you help and support
>
ok thanks a lot for your support
2011/9/27 Shaun Ruffell
> On Tue, Sep 27, 2011 at 09:01:32AM +0000, salaheddine elharit wrote:
> > thank you evryone for you help and support
> >
> > i make this command lspci -d d161:* and i get :
> >
> > 09:02.0 Communication
hello list
i have configured a sip account in order to do an outbound calls and i want
to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHE
File to play when call begins.
* LIMIT_WARNING_FILE File to play as warning if 'y' is
defined.
The default is to say the time
remaining.
but i don't understand what i can do to solve this
thanks
2011/9/28 Paul Belanger
>
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah
> have a look at the following:
> "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left,
> repeated every 'z' ms) Only 'x' is required,
un...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Wednesday, September 28, 2011 1:23 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Limit outbond calls duration to
t;
>
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Wednesday, September 28, 2011 1:23 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
ok thanks for your response i will try that and i will update you as soon as
i have any result
best regards
2011/9/29 A J Stiles
> (top-posting mess fixed the lazy man's way .)
>
> On Thursday 29 September 2011, salaheddine elharit wrote:
> > ok thanks it's work fi
Hi
Thanks everyone for your help and support all works perfectly
Best Regards
2011/9/29 salaheddine elharit
> ok thanks for your response i will try that and i will update you as soon
> as i have any result
>
> best regards
>
> 2011/9/29 A J Stiles
>
>> (top-
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten => 0678XX,1,Set(CALLERID(number)=520XX)
exten => 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQ
s in /var/lib/asterisk/moh1?
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, October 04, 2011 12:49 PM
> *To:* Asterisk Users Mailing List
> call and directory couldn't play anything. So I fixed that by granting
> directory permissions.
>
>
> On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> Hi
>>
>> yes i have noticed the same result when i
Hello list
i have one question related to meetme,i have to providers with the first one
i put the number with 9 digit 520XX and all works without issue, with
the second i put just the last 3 numbers 500 with meetme there is nothing
but when i put the last 3 numbers like below i can call m
are using pattern matching and not using the right syntax
> like that.
> exten => _520,1,answer
> like that.
> On 5 Oct 2011 21:47, "salaheddine elharit"
> wrote:
> > Hello list
> >
> >
> >
> > i have one question related to meetme,i h
hello list
i have configured the voicemail in my server asterisk 1.4 i can use it
without issue ,i have a question
i want to receive an email in my address email when there is no response
from 270 after 10 s
could you please verify the code below and tell me what is wrong
thanks and reg
thanks for your response
itry this but i didn't recive any email,also if there is a way to recive a
SMS in my mobile 0678XX
regards
2011/10/18 Ishfaq Malik
> On Tue, 2011-10-18 at 12:26 +, salaheddine elharit wrote:
> > hello list
> >
> >
> >
> &
eiving email to trigger
> an SMS when a particular type of message is recieved in inbox.(We used this
> one)
> c- Configure kennel to Send out SMSs and write an integration bash
> script to be called after the voicemail application.
>
>
>
> On Tue, Oct 18, 2011 at 8:4
hi list
i have an issue related to queues.conf
i have configured the code like below
extensions.conf
[default]
exten => 800,1,AgentLogin()
exten => 666,1,Answer()
exten => 666,2,Queue(hotline)
#include "aheeva_dialplan.conf"
===
queues.conf
[hotline]
member => Agent
class 'default', on Zap/14-1
-- Channel 1/14, span 1 got hangup, cause -1
-- Stopped music on hold on Zap/14-1
== Spawn extension (default, 560, 2) exited non-zero on 'Zap/14-1'
-- Hungup 'Zap/14-1'
please advice
2011/10/20 Warren Selby
> On
why for the first provider we put all the number in
extensions .conf but for the second provider we put just the last 3 numbers
thanks for your help
2011/10/21 Warren Selby
> On Fri, Oct 21, 2011 at 4:50 AM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>
hello list
i have installed asterisk 1.8.7.1 and i have configured 2 account for sip
in order to do internal call
when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from
223 to 222
but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to
snom320 but the issue i c
thank you so much all works without issue now
2011/10/31 Christian Gansberger
> Hello,
>
> You have to disable RTP-Encryption on your Snom under Identity, RTP.
> It is set to on per default.
>
>
> On 31 October 2011 13:33, salaheddine elharit
> wrote:
>
hello list
i have asterisk 1.4 installed and i want to use CDR mysql during the
installation i didn’t check the cdr mysql with make menuselect
my question : i want to check this option now after the installtion and
configuration of all options but he asks me to do. /configure before to use
issue
thanks for your response
2011/11/21 Ryan Wagoner
> On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit
> wrote:
> > hello list
> >
> >
> >
> > i have asterisk 1.4 installed and i want to use CDR mysql during the
> > installation i d
hello list
i have asterisk 1.4 with and i have one card diguim (E1) with 2 providers
i have noticed by an error related to the first provider some times i can
not call the numbers of of this provider
but with the second one there is no issue alos i cal call the internal
extension without issue
f
py cdr_mysql.so to /usr/lib/asterisk/modules and load the module from
> CLI
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Monday, November 21
hello list,
i have created one menu like below all work without issue, what i want to
do is ,
when the customer press 3 in menu context exten => 3,1,Goto(support,s,1)
i want to stok this variable (3) in database or file instead to go to
support context
thanks for your help and support
best re
u,s,1)
2011/11/25 Dale Noll
>
>
> On 11/25/2011 09:32 AM, salaheddine elharit wrote:
>
>> hello list,
>> i have created one menu like below all work without issue, what i want to
>> do is ,
>> when the customer press 3 in menu context exten =>
>> 3,
thank you so much i will test this option and i will update you
2011/11/25 Dale Noll
> **
> On 11/25/2011 10:48 AM, salaheddine elharit wrote:
>
>
> thanks for your response
>
> i use mysql like a database and my question when the customer press 3 in
> context menu i wa
lation-support)
exten => s,3,Goto(menu,s,1)
exten => s,4,Playback(${sounds_path}relation-support)
exten => s,5,Queue(default)
exten => t,1,Hangup()
2011/11/25 Dale Noll
> **
> On 11/25/2011 10:48 AM, salaheddine elharit wrote:
>
>
> thanks for your response
>
>
hello list
i have one question how to check if the app_mysql module is compiled and
loaded in my system if not how can i do in order to install it and load it
i use asterisk 1.4
thanks and regards
--
_
-- Bandwidth and Colocatio
t; ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Monday, November 21, 2011 8:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [a
e show like sql
> ls -l /usl/lib/asterisk/modukes/*sql*
>
>
> 2011/11/29 mahesh katta
>
>> yum info mysql* check with this
>>
>> Reg:
>> Mahi.
>>
>>
>> On Tue, Nov 29, 2011 at 3:20 PM, salaheddine elharit <
>> salah.elharit...@g
> ./configure.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, November 29, 2011 5:37 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial
-on.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, November 29, 2011 8:45 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re
Nicholas
> Try running
>
> ./configure –with-mysql=/usr
>
> Make menuselect
>
> ** **
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> e
risk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, November 29, 2011 9:20 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] CDR mysql with aster
source you want for this one as I don’t use mysql or any
> databases with my asterisk (prefer to keep things as simple as possible).*
> ***
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddi
ber 2011, salaheddine elharit wrote:
> > after
> >
> > ./configure –with-mysql=/usr
> >
> > Make menuselect
> >
> > ** **same result but i found in module embedding the below options i
> check
> > the cdr and i get the same result
> >
in no way replace anything you currently have or use.
>
> Regards
>
> Ish
>
> On Tue, 2011-11-29 at 17:30 +, salaheddine elharit wrote:
> > i use centos 5.5 if i install mysql-devel i can still use the version
> > of mysql installed now in my server because i use it wi
thank you so much for you help,i have flowed your email and installed
thesesadd-ons all
works perfectly i can store the phone_number of the Customer ,now i can do
what i want :)
thanks every one for your support J
2011/11/30 Dale Noll
> On 11/28/2011 08:24 AM, salaheddine elharit wr
ease give me the syntex
best regards
2011/11/30 salaheddine elharit
> thank you so much for you help,i have flowed your email and installed
> theses add-ons all works perfectly i can store the phone_number of the
> Customer ,now i can do what i want :)
>
>
>
> thanks every
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll
>
> On 11/30/2011 11:13 AM, salaheddine elharit wrote:
>
>> i have last question regarding this thread
>> with exten => 3,n,MYSQL(Query result
Hello List
coud you please show me how to get the RECORD_ID for all outbond calls, i
use asterisk 1.4 with database mysql
thanks and regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asteri
Hi Bilal
in my case i use an IVR menu using asterisk 1.4 an i can store the number
of the customer in my database and after i can select
the phone number and the date_time of calling i use mysql
you must change database login password with yours and also the name of
table
regards
exten => 500x
Hello,
in order to play the music in asterisk like a MusicOnHold you can convert
it from MP3 to GSM
Regards,
2010/12/2 Steve Edwards
> On Thu, 2 Dec 2010, Ernie Dunbar wrote:
>
> > I have some MP3 files that play well in any MP3 player I throw at them,
> > but when I try to make a MusicOnHold
Hello All,
i have asterisk installed in my call centre without any issue I would like
to ask you some questions related to services.
i want to start asterisk and httpd and aheevacti automatically when the
server centos reboot or shutdown
becouse i must start all services manually (service
ok thank you so much for your help
2010/12/20 Doug Lytle
> salaheddine elharit wrote:
>
>> becouse i must start all services manually (service asterisk start
>> ,service httpd start
>>
>
> chkconfig httpd on
> chkconfig asterisk on
>
> Doug
>
>
>
are ON ,
and when i reboot the server i found that the service httpd is off with
command "service httpd status" and service asterisk status
please advice
Best Regards,
2010/12/20 salaheddine elharit
> ok thank you so much for your help
>
> 2010/12/20 Doug Lytle
>
>
*Hello* all,
i have asterisk installed in our call centre and I have all the clients
conversation saved in this file
/usr/apache-tomcat-5.5.17/webapps/aheevaccs/recordings/nosales
i have created i php code in order to read this files from server
when i put the file in www folder i can re
yes i want to know how can i do in order to read this files using apche
2011/1/17 Steve Edwards
> On Mon, 17 Jan 2011, salaheddine elharit wrote:
>
> i have asterisk installed in our call centre and I have all the clients
>> conversation saved in this file
>>
>>
Thank you so much for your response I will try this operation and I will
update you as soon as I have any result
2011/1/18 A J Stiles
> On Tuesday 18 Jan 2011, salaheddine elharit wrote:
> > yes i want to know how can i do in order to read this files using apche
>
> Either make
Hello All,
I have asterisk installed in our call center and i want to know how to do in
order to save all the calls (inbound and outbound) if there is any tool
Thanks in advance
Kind Regards.
--
_
-- Bandwidth and Colocati
Hello all
I have a small issue with some mobiles numbers when I call these numbers
using asterisk I have all the time answer machine. But when I call these
numbers using my mobile or another phone there is no problem.
Any help will be appreciated
--
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6XXX" ${AH_PHONE_NUMBER})})
exten => _OUT.,n,GotoIf($["${match}" = "1"]?rien)
thanks and regards
2011/2/14 Danny Nicholas
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.
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