t;
> Thanks a lot
>
>
> 2011/3/1 Danny Nicholas
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, March 01, 2011 10:35 AM
to:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, March 01, 2011 10:35 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] records inbound and outbound calls
>
>
>
> Hello List
Hello List
i have asterisk installed in our call centre i have configured the snom
phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com
i have just one question how can i do in order to record all the calls
automatically in our server
Thanks and regards
--
___
evaCCS-3.1.0_Build-11629 currently
running on srvradio (pid = 24818)
Verbosity is at least 3
[Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite:
Call from '106' to extension '1018' rejected because extension not found.
srvradio*CLI>
thank you for your h
Hello,
i have asterisk installed and i have configured a client iax and sip without
any issue, when i call a internal extension sip from iax there is no problem
but when i call a iax extension from sip extension the result is KO(wrong
number)
any help please
thanks and Regards
--
__
t;]?entno)
>
> exten => 999,n,SayDigits(${digito})
>
> exten => 999,n,Set(DB(block/${digito}=${digito}))
>
> exten => 999,n,BackGround(vm-goodbye)
>
> exten => 999,n,Hangup(${HANGUP_CAUSE})
>
>
> --
>
> *From:* asteris
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.
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
--
___
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
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
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
>>
>>
*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
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
>
>
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
>
>
>
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
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,
i had the same issue when using x-lite when i verify i found that the issue
is related to configuration of x-lite i change the value in x-lite option
and now there is no issue all function good
Hope it can help you
2010/6/30 Anahi Ludueña
> Hi people,
> we have some extensions which a
hello
you must to do a configuration of yor sip.conf
like that
[the login of sip]
type=friend
context=default
secret=(the password of sip )
host=dynamic
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
Regads
2010/7/5 Pezhman Lali
> add the a2billing configurations to
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Friday, June 18, 2010 12:56 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users
nt, since some installs/os’es lend
> themselves to memory leaks.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Friday, June 18,
do you have any tool in order to check what happened in asterisk during the
hangs of calls
2010/6/18 salaheddine elharit
> Hello,
>
>
>
> I have a problem in Asterisk 1.4 each day I need to restart *asterisk
> service asterisk* restart in order to unblock the calls
>
Hello,
I have a problem in Asterisk 1.4 each day I need to restart *asterisk
service asterisk* restart in order to unblock the calls
My question how can I do in order to check the issue, and if there is any
tool or log?
Thanks and regards.
--
_OUT.,6,AHEventsProxy(MSG_TYPE_CALL_SIT:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten => _OUT.,7,Goto(9)
exten =>
_OUT.,8,AHEventsProxy(${DIALSTATUS}:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten => _OUT.,9,NoOp()
thanks a lot
2
Hello everyone,
>
any help please
>
>
> I have asterisk installed in our call centre with aheeva platform and
> centos linux,
>
>
>
> We have 2 access provider I have configured the
> etc/asterisk/extensions.conf in order to do the routing of calls
>
>
>
> exten => _0612.,1,Set(CALLERID(number
Hello everyone,
I have asterisk installed in our call centre with aheeva platform and centos
linux,
We have 2 access provider I have configured the etc/asterisk/extensions.conf
in order to do the routing of calls
exten => _0612.,1,Set(CALLERID(number)=520460587)
exten => _0612.,n,Dial(Zap
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
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
> >
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
--
___
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
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
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
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