hi
you can try this link
http://zaf.github.io/asterisk-googletts/
2015-08-26 19:15 GMT+01:00 Tech Support aster...@voipbusiness.us:
All;
I have a customer who is looking for a good speech to text solution,
either open source or reasonably priced commercial product, I’m open to
what about
exten = s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
regards
2015-04-08 5:45 GMT+00:00 Dmitriy Serov serov@gmail.com:
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1.
.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com
:
hello list
i need your help
please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and
:35 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same provider in my asterisk
i can call all numbers just the numbers are configured
GMT+00:00 A J Stiles asterisk_l...@earthshod.co.uk:
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
, that's great news.
On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit
salah.elharit...@gmail.com wrote:
i noticed that when i active the voicemail in the IP-phone where the
number 0033149xx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP
Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i
configure the trunk directly in x-lite i
i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xx == Begin MixMonitor Recording
SIP/101-010d
--
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i configure
the trunk directly in x-lite i can contact this number without issue
below the cli
==
thank you so much Carlos ;the issue has been solved
Best Regards.
2015-03-12 18:40 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi
007100 for exemple i spy another agnet 102 or 103
any help please
thanks and regards
2015-03-12 10:30 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten
, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from
hello list,
i have created a queue with and i have a question related to musiconhold
f there is any way to set the musiconhold just for caller not for agent
logged in the queue
thanks and regards.
--
_
-- Bandwidth and
hello liste
i have creat i trunk sip and inboun route for inbound calls the issue whe i
use the DID in inboud route i have a error No DID or CID Match.
but when i leave this DID field blank i can route the call without any issue
how can ido in order to use DID in route inboud i use elastix
Hello list,
i have installed elastix 2.4.0 with call center model and i have
created an Outgoing
Calls https://192.168.1.251/index.php?menu=outgoing_calls my question i
want to know the name of the tbale where the csv file is uploaded in order
to do some works.
NB: i found the cdr table in
hello list,
i have a question i don't know if there is any possibility to stop asterisk
using a call for exp:
when i call a number 0522xx i want to excute a script or any idea to
stop asterisk automatically
i use asterisk 1.4.43
NB: with mysql using a database i can insert into table using
thanks a lot it works correctly
2014-04-07 12:08 GMT+00:00 Andres and...@telesip.net:
On 4/7/14, 4:53 AM, Salaheddine Elharit wrote:
hello list,
i have a question i don't know if there is any possibility to stop
asterisk using a call for exp:
when i call a number 0522xx i want
hello,
try to use failed instead of h
exten = failed,1,
best regards.
2014-02-18 9:09 GMT+00:00 Ishfaq Malik i...@pack-net.co.uk:
What version of asterisk are you using?
Ish
On 17 February 2014 20:49, Mike Diehl mdiehlena...@gmail.com wrote:
Hi all,
I'm trying to build a fax
thanks and regards
2014-02-05 Larry Moore lmo...@omninet.net.au:
On 6/02/2014 2:21 AM, Salaheddine Elharit wrote:
thanks for your response ,
i test this solution but the issue still the same
any other solution
thanks and regards
2014-02-04 Steve Edwards asterisk@sedwards.com
thanks for your response ,
i test this solution but the issue still the same
any other solution
thanks and regards
2014-02-04 Steve Edwards asterisk@sedwards.com:
On Tue, 4 Feb 2014, Salaheddine Elharit wrote:
i have asterisk 1.4.43 installed and i want to configure the auto-answer
hello list,
i have asterisk 1.4.43 installed and i want to configure the auto-answer
exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0)
exten = 506,n,Dial(SIP/105)
when i call the 506 the SIP/105 still ringing, i have snom 320 and i have
set the Auto Answer Indication: on
i test with
hello list,
i have created a callfiles with my asterisk 1.4.43 like:
Channel: SIP/watara/06
MaxRetries: 10
RetryTime: 5
WaitTime: 20
Context: mycontext
Extension: s
Priority: 1
extensions.conf
mycontext
exten = s,1,Ringing()
exten = s,n,Playback(hello-world)
exten = s,n,Dial(SIP/105)
hello list
i have create i trunk Sip between 2 servers in the same network
when i call a number (inbound calls) in the first server i can forward this
number to my sip 222 in the second server
exten = 0522xx,1,Dial(SIP/222@trunk_created,30)
my question if there is any possibility
-info.org
See also the [stdexten] section of extensions.conf.sample
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
Sent: Thursday, December 19, 2013 1:32 PM
To: Asterisk Users Mailing List
i attached file my dialplan
2013/12/20 Salaheddine Elharit salah.elharit...@gmail.com
in attached file my dialplan
thanks and regards
2013/12/20 Eric Wieling ewiel...@nyigc.com
You must write dialplan code to do what you want. Assuming you are not
using a GUI with Asterisk, post
I have this scenario
In the first server 192.168.5.100 I have asterisk installed 1.4.43 and one
diguim card with 2 ports: in the first port connection for the provider X :
the second port of diguim card the connection of the provider Y
In the second server (the same configuration)
...@lists.digium.com] On Behalf Of Salaheddine Elharit
Sent: Thursday, December 19, 2013 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] send the calls from to servers
I have this scenario
In the first server 192.168.5.100 I have asterisk installed 1.4.43
hello johan,
i use Authenticate and i get what i want thank you so much for your help :)
exten = 600,1,Ringing(2)
exten = 600,n,Answer
exten = 600,n,Authenticate(1234)
exten = 600,n,Goto(home,s,1)
2013/12/5 Steve Edwards asterisk@sedwards.com
On Thu, 5 Dec 2013, Salaheddine Elharit wrote
--
From: Salaheddine Elharit salah.elharit...@gmail.com
Date: 2013/11/29
Subject: Re: [asterisk-users] issue with speech in IVR
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
hello
i add the following in chan_dahdi and the issue has been
:36, Salaheddine Elharit wrote:
hi
i follow your dialplan but the issue still the same ican't stop the speech
and go to another context
any other idea please
best regards .
It sounds as thgough the DTMF tones are not being sent in a way that
Asterisk is seeing .
What type
echocancel = no
dtmfmode = auto
Mitul
On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote:
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use
the keypad to control the playback
-Original Message-
From: Salaheddine Elharit salah.elharit
: Salaheddine Elharit salah.elharit...@gmail.com
Date: 2013/11/29
Subject: Re: [asterisk-users] issue with speech in IVR
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
hello
i add the following in chan_dahdi and the issue has been solved thanks a
lot
,1)
2013/11/28 Paul Belanger paul.belan...@polybeacon.com
On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
hello list
i have an IVR menu in asterisk 1.4
like below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar
hi
i follow your dialplan but the issue still the same ican't stop the speech
and go to another context
any other idea please
best regards .
2013/11/28 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 27 November 2013, Salaheddine Elharit wrote:
hello list
i have an IVR menu
On Thu, Nov 28, 2013 at 8:36 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hi
i follow your dialplan but the issue still the same ican't stop the
speech and go to another context
any other idea please
best regards .
My guess is that your DTMF tones are not reaching
hello list
i have an IVR menu in asterisk 1.4
like below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}music1)
exten =
Hello list
i have an issue with my dahdi_channels.conf
in span 1 when i use it like below i can do my outband calls without issue
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel = 17-31
context = default
group
callerid=52xx
immediate=no
channel = 32-46,48-52
thanks and regards
2013/10/31 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 31 October 2013, Salaheddine Elharit wrote:
Hello list
i have an issue with my dahdi_channels.conf
in span 1 when i use it like below i can do my
thanks for your response i will swap the cables and i will update by the
result
best regards
2013/10/31 Tony Mountifield t...@softins.co.uk
In article
cahexamsp4nenuntymuzwjgep69v+7rb7ekbyzsalmbm+zyo...@mail.gmail.com,
Salaheddine Elharit salah.elharit...@gmail.com wrote:
below
etc
is not available.
could you please help me
thanks and regards
2013/10/24 Salaheddine Elharit salah.elharit...@gmail.com
ok thanks for your comment i really appreciate it
best regards
2013/10/23 Russ Meyerriecks rmeyerrie...@digium.com
On Wed, Oct 23, 2013 at 11:27 AM, Salaheddine Elharit
ok thanks for your comment i really appreciate it
best regards
2013/10/23 Russ Meyerriecks rmeyerrie...@digium.com
On Wed, Oct 23, 2013 at 11:27 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hi
the issue has been solved after change the span from span
=1,1,0,ccs,hdb3
hi
the issue has been solved after change the span from span =1,1,0,ccs,hdb3
to span =1,0,0,ccs,hdb3
thanks for everyone
2013/10/22 Salaheddine Elharit salah.elharit...@gmail.com
2013/10/22, A J Stiles asterisk_l...@earthshod.co.uk:
On Tuesday 22 October 2013, Salaheddine Elharit wrote
or beyond.
Also, CLI says 1.4.43, your message says 1.4.32 ???
Some examination of chan_dahdi and your dialplan would help someone give
you some assistance.
Is this a fresh install, or one that has been working for years?
What Digium card?
John Novack
Salaheddine Elharit wrote:
i
2013/10/22, A J Stiles asterisk_l...@earthshod.co.uk:
On Tuesday 22 October 2013, Salaheddine Elharit wrote:
hello yes this is a fresh install
[trunkgroups]
trunkgroup = 1,16
spanmap = 1,1,1
[channels]
#include dahdi-channels.conf
context=default
hidecallerid=no
callwaiting=yes
i need your help regarding some issue related to the outband calls
i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2
ports
when i try to call my phone number all time i receive message busy number
this error just with g1.
with g2 there is no problem i can call without
thanks for your response
with the code below i can't get the extenssions 223
exten = 529,1,Answer()
exten =
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
i can get my number only with
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten =
i have Create a h extension and all works without issue .thank you so
much for your help and support i really appreciate it.
2013/7/31 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached
RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with
the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf
)
exten = 534,102,NoOp(We reached step 102)
2013/7/31 Joshua Colp jc...@digium.com
A J Stiles wrote:
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from
,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
thanks and regards
2013/7/25 Salaheddine Elharit salah.elharit...@gmail.com
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles
(answered),NoOp(Call was answered)
any help please
2013/7/26 A J Stiles asterisk_l...@earthshod.co.uk
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call
Hello list,
i need your help about the IVR please
i have asterisk 1.4 installed and i configure an IVR like below
exten = 529,1,Ringing()
exten = 529,n,Wait(4)
exten = 529,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten =
chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
any help please
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
i have asterisk 1.4 installed and i configure an IVR like
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10
hello
if you have just some numbers to block you can use the below code in your
dial plan
exten = 5xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 5xx,n,GotoIf($[${CALLERID(num)}=0661xx ]?3:4)
exten = 5xx,n,hangup
exten = 5xx,n,Dial(SIP/223, 30)
2013/6/17 A J Stiles
hello list ,
i want to use meetme with asterisk1.4 i check in this forum and i found
this code :
exten = 508,1,MeetMe(1000,ipdM)
when i use this code in my server i can say my name and i press 1 in order
to enter in the conference ; but i want to asks the customer to press an
number and
hello list,
i need your help please regarding send mail i use astreisk 1.4;
i try to send mail when no response like below
exten = 5xx,1,Dial(SIP/223, 10)
exten = 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed'
myadresseem...@gmail.com)
when i launch the CLI i found :
You have
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
Elharit
*Sent:* Thursday, May 30, 2013 8:07 AM
*To:* **Asterisk Users Mailing List - Non-Commercial Discussion**
*Subject:* [asterisk-users] how
hello ,
thanks alex for your help and support the scenario is correct.
i will try to follow your suggestion and i will update you asap
thank you again for your explication i really appreciate it
2013/5/31 Alex Villacís Lasso a_villa...@palosanto.com
El 31/05/13 09:21, Salaheddine Elharit
Hello
i want to luanch an URL in my PC when i call a number like below
exten = 066104,1,Set(CALLERID(number)=52xxx)
exten = 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten
=
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten =
hi
You can download a tarball of the release here:
http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz
2013/5/11 Andrew Colin and...@vsave.co.za
I thought he said rhel 6.3
Sent from my iPhone
On 11 May 2013, at 2:48 PM, Asghar Mohammad
can use
506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want
to do failover the check Dial status and gotoif dialstatus = NO ANSWER or
what ever you need.
On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello list,
i need your
hello list,
i need your help about cdr ,i have installed the module cdr in my asterisk
1.4 .
for the inbound calls when i call my sip exten like below :
exten = 506,1,Dial(SIP/223, 10)
exten = 506,n,Dial(SIP/276, 10)
in CDR i have just one line with SIP /276 the last line but there is
no
thanks i verify but i don't understanding if can someone give me an example
best regards
2013/5/9 Ishfaq Malik i...@pack-net.co.uk
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
hello list,
i need your help about cdr ,i have installed the module cdr in my
asterisk
())
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
thanks and regards
2011/12/1 salaheddine elharit salah.elharit...@gmail.com
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn
-15,17-31
group=2
callgroup=2
switchtype=qsig
signalling=pri_net
callerid=mycallerid
immediate=no
channel = 156-170
channel = 172-176
channel = 125-139
channel = 141-155
thanks and regards
2013/3/27 Yves A. yves...@gmx.de
Am 26.03.2013 17:57, schrieb Salaheddine Elharit:
Hello
. The configure script
gives me the option unused for any port. Maybe your configure script
offers you the same option.
Am 27.03.2013 11:54, schrieb Salaheddine Elharit:
Hi
i use 2 digium cards 1 card with 2 ports and the second card with 4 ports
but actually i use just the span 1 and span
to help you find the
answer yourself... it can
be very frustrating sometimes, but for me, thats all i can tell about.
regards,
yves
Am 27.03.2013 13:06, schrieb Salaheddine Elharit:
thank you for your help ,but which configure script and when i can find
this script ? in etc/asterisk
Hello,
i have all the time this warning i use asterisk 1.4 all works without
issue i don't have any problem (i can use the inbound and outbound calls
without issue)
i just want to know what is this WARNING
thanks and regards
WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels
ok thanks for your help and support i really appreciated
2013/3/26 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote:
hello list,
i have a question related to zapata.conf,if i do any change in
zapata.conf
i must restart asterisk
hello list,
i have a question related to zapata.conf,if i do any change in zapata.conf
i must restart asterisk or just i restart zapata ,and how to do .
“service zaptel restart” or there is any other command
Thanks and regards
--
22, 2013 at 7:14 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2
applied is to stop asterisk, reload the
driver and than start asterisk again.
regards,
yves
btw..:
zaptel ist outdated... you should definitely upgrade using dahdi drivers...
Am 25.03.2013 11:44, schrieb Salaheddine Elharit:
hello list,
i have a question related to zapata.conf,if i
Wieling ewiel...@nyigc.com
Service asterisk stop
Service zaptel restart
Service asterisk start
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
Sent: Monday, March 25, 2013 11:04 AM
channel when dialing...
if you´re to afraid to do it... then leave it as it is and follow the
ntars-maxime (never touch a running system)...
regards,
yves
Am 25.03.2013 16:15, schrieb Salaheddine Elharit:
thank you so much
fo the upgrade from zptel to dahdi, if there is any possibility
=
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2
Hello bharat,
ok thank you so much for your help and support now i understand :)
2013/3/22 Bharat Lalcheta bharatlalch...@gmail.com
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok
and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works
without problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
want to active the round-robin for span 2 and 6) in order to activate the
WIMAX and FH
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4
,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
Elharit
*Sent:* Wednesday, February 20, 2013 10:33 AM
*To:* **Asterisk Users Mailing List - Non-Commercial Discussion**
*Subject:* [asterisk-users] issue with inbound
thank you so much for your response the issue was solved after using
http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz
best regards
2013/2/15 Russ Meyerriecks rmeyerrie...@digium.com
/usr/src/dahdi-linux-2.6.1/drivers/dahdi/xpp/xdefs.h:152: error:
I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210
with 2 port E1.
now i bought another card Diguim TE410 and I want to add it
the current configuration : connection (WIMAX) from the first ISP and
connection (fiber optic) from the secend ISP.
the desired configuration :
, Salaheddine Elharit
salah.elharit...@gmail.com ha scritto:
I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210
with 2 port E1.
now i bought another card Diguim TE410 and I want to add it
the current configuration : connection (WIMAX) from the first ISP and
connection (fiber
Hello list
could you please help me about one question.
i have asterisk 1.4 installed, i configure the inbound call in my asterisk
like below.
exten = 520xx,1,Dial(SIP/224, 30).
when the customer call my number (520xx) the sip phone 224 works
without issue
my problem i
thanks danny
i think i didn’t explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to block
it to call my number 520xx .
2013/1/14 Danny Nicholas da...@debsinc.com
Exten = _0666XX,1,answer()
Exten =
)} = 0666XX ]?3:4)
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
2013/1/14 Salaheddine Elharit salah.elharit...@gmail.com
thanks danny
i think i didn’t explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to block
it to call my
] *On Behalf Of *Salaheddine
Elharit
*Sent:* Monday, January 14, 2013 10:51 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] block one number in incoming calls
** **
hi Zohair Raza
** **
thanks for your replay but this script will allow just
:
On Monday 14 January 2013, Salaheddine Elharit wrote:
i think i didn’t explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to
block
it to call my number 520xx .
Use something like
Exten = _520X./0666XX,1,Answer()
Exten = _520X
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 =
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
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn...@wi.rr.com
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid
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 dn...@wi.rr.com
On 11/28/2011 08:24 AM, salaheddine
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