Re: [asterisk-users] Anyone doing speech to text?

hi you can try this link http://zaf.github.io/asterisk-googletts/ 2015-08-26 19:15 GMT+01:00 Tech Support : > 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 > suggestions. > > Thanks; >

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

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 : > Hi, Andrew. > > You are trying to solve two tasks: definition through what line the call > came and a beautiful display of this information. > 1. definition throug

Re: [asterisk-users] call between snom 300 and aastra 6731i

") in new stack -- Executing [s@macro-outisbusy:4] Playback("SIP/300-0192", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack [2015-03-27 18:35:19] WARNING[350][C-00f3]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in an

Re: [asterisk-users] call between snom 300 and aastra 6731i

please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit : > 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 > >

[asterisk-users] call between snom 300 and aastra 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 aa

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

e ip and port 2015-03-25 13:47 GMT+00:00 A J Stiles : > ** 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

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

AM, Salaheddine Elharit > 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 co

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

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 in

Re: [asterisk-users] outbound calls

t'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 >&g

Re: [asterisk-users] outbound calls

cognize >> that is was coming from you. You might compare the SIP INVITE coming from >> Asterisk to the one from Z-Lite and see where the differences are. >> >> >> >> On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit < >> salah.elharit...@gmail.com&

Re: [asterisk-users] outbound calls

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 -- SIP/FD-

[asterisk-users] outbound calls

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 == U

Re: [asterisk-users] chanspy for group extension

thank you so much Carlos ;the issue has been solved Best Regards. 2015-03-12 18:40 GMT+00:00 Salaheddine Elharit : > 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 : > >> Hi, &

Re: [asterisk-users] chanspy for group extension

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 : > Hi, > > Le 12/03/2015 17:28, Salaheddine Elharit a écrit : > >> hello list, >> >> i use the code below >> >> [macro-ch

Re: [asterisk-users] chanspy for group extension

8910) but when i do 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 : > thank you so much it work > you must add 1 like below > > [app-chanspy] > exten => _0071XX,*1,*Macro(chanspy,1234) >

Re: [asterisk-users] chanspy for group extension

thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez : > On 3/11/15 12:48 PM, Salahed

[asterisk-users] chanspy for group extension

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 fro

[asterisk-users] set musiconhold only for caller

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 Colocati

[asterisk-users] issue with inbound route

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" E

[asterisk-users] asterisk and elastix

Hello list, i have installed elastix 2.4.0 with call center model and i have created an 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 i

Re: [asterisk-users] how to stop asterisk using a call

thanks a lot it works correctly 2014-04-07 12:08 GMT+00:00 Andres : > 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

[asterisk-users] how to stop asterisk using a call

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

Re: [asterisk-users] h extension isn't processed after call file finishes.

hello, try to use failed instead of h exten => failed,1, best regards. 2014-02-18 9:09 GMT+00:00 Ishfaq Malik : > What version of asterisk are you using? > > Ish > > > On 17 February 2014 20:49, Mike Diehl wrote: > >> Hi all, >> >> I'm trying to build a fax relay mechanism where faxes

Re: [asterisk-users] auto-answer call

2) exited non-zero on 'DAHDI/13-1' -- Hungup 'DAHDI/13-1' and the call hungup when i use the Dial the sip/105 still ringing thanks and regards 2014-02-05 Larry Moore : > On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: > >> thanks for your respo

Re: [asterisk-users] auto-answer call

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 : > On Tue, 4 Feb 2014, Salaheddine Elharit wrote: > > i have asterisk 1.4.43 installed and i want to configure the auto-answer >> &g

[asterisk-users] auto-answer call

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

[asterisk-users] callfiles.call

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

[asterisk-users] go to context from server 1 to server 2

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 t

Re: [asterisk-users] send the calls from to servers

i attached file my dialplan 2013/12/20 Salaheddine Elharit > in attached file my dialplan > > thanks and regards > > > > > 2013/12/20 Eric Wieling > >> You must write dialplan code to do what you want. Assuming you are not >> using a GUI with Asteris

Re: [asterisk-users] send the calls from to servers

ives > and voip-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

Re: [asterisk-users] send the calls from to servers

.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

[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 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) 192.168.5.2

Re: [asterisk-users] issue with speech in IVR

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 > On Thu, 5 Dec 2013, Salaheddine Elharit wrote:

Re: [asterisk-users] issue with speech in IVR

ath=/var/lib/asterisk/sounds/) exten => s,n,Background(${sounds_path}error any example would be appreciated 2013/11/29 Mitul Limbani > Sounds cool, I suspected the echo cancel situation, these are usually > issue even for FAX communication on dahdi. > > Mitul > > > On F

[asterisk-users] Fwd: issue with speech in IVR

: Salaheddine Elharit 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 for your help and s

Re: [asterisk-users] issue with speech in IVR

ocancel = no > dtmfmode = auto > > Mitul > On Nov 29, 2013 1:42 PM, 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

Re: [asterisk-users] issue with speech in IVR

ess 1 for xx 1 press 2 for yyy i sotre his phone number and his choice in my database for me the issue the customer he can nto wait the speech of unless and finished . best regards i use a diguim card with PRI 2013/11/29 A J Stiles > On 28/11/13 15:36, Salaheddine

Re: [asterisk-users] issue with speech in IVR

> 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 >&g

Re: [asterisk-users] issue with speech in IVR

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 > On Wednesday 27 November 2013, Salaheddine Elharit wrote: > > hello list > > > > i have a

Re: [asterisk-users] issue with speech in IVR

X,2,Goto(project,s,1) 2013/11/28 Paul Belanger > 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) >>

[asterisk-users] issue with speech in IVR

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(${sounds_path}music1) exten => s,n,Background(${sounds_path}m

Re: [asterisk-users] issue with dahdi_channels.conf

thanks for your response i will swap the cables and i will update by the result best regards 2013/10/31 Tony Mountifield > In article < > cahexamsp4nenuntymuzwjgep69v+7rb7ekbyzsalmbm+zyo...@mail.gmail.com>, > Salaheddine Elharit wrote: > > > > below &g

Re: [asterisk-users] issue with dahdi_channels.conf

hannel => 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=52xx immediate=no channel => 32-46,48-52 thanks and regards 2013/10/31 A J Stiles > On Thursday 31 October 2013, Salaheddine Elharit wrote: > > Hello list > > > > > > i have a

[asterisk-users] issue with dahdi_channels.conf

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 gro

Re: [asterisk-users] issue after install dahdi

uested channel 1/2 is not available. could you please help me thanks and regards 2013/10/24 Salaheddine Elharit > ok thanks for your comment i really appreciate it > > > best regards > > > 2013/10/23 Russ Meyerriecks > >> On Wed, Oct 23, 2013 at 11:27 AM,

Re: [asterisk-users] issue after install dahdi

ok thanks for your comment i really appreciate it best regards 2013/10/23 Russ Meyerriecks > 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

Re: [asterisk-users] issue after install dahdi

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 > 2013/10/22, A J Stiles : > > On Tuesday 22 October 2013, Salaheddine Elharit wrote: > >> hello yes this

Re: [asterisk-users] issue after install dahdi

2013/10/22, A J Stiles : > 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 &g

Re: [asterisk-users] issue after install dahdi

8.current 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 N

[asterisk-users] issue after install dahdi

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

Re: [asterisk-users] (no subject)

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 wi

[asterisk-users] (no subject)

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 => 529,n

Re: [asterisk-users] asterisk and IVR

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 > On Wednesday 31 July 2013, Salaheddine Elharit wrote: > > hi > > > > i use the code below but i didn't g

Re: [asterisk-users] asterisk and IVR

,1) exten => 534,n(answered),NoOp(Call was answered) exten => 534,102,NoOp(We reached step 102) 2013/7/31 Joshua Colp > A J Stiles wrote: > >> * PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE * >> >> On Wednesday 31 July 2013, Salaheddine E

Re: [asterisk-users] asterisk and IVR

e' (language 'en') -- Channel 0/23, span 1 got hangup request, cause 16 == Spawn extension (home, s, 2) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' 2013/7/26 A J Stiles > * THIS IS NOT WHERE YOUR RESPONSE GOES * > > On Friday 26

Re: [asterisk-users] asterisk and IVR

xten => 534,n,Goto(home,s,1) exten => 534,n(answered),NoOp(Call was answered) any help please 2013/7/26 A J Stiles > * THIS IS NOT WHERE YOUR RESPONSE GOES * > > On Friday 26 July 2013, Salaheddine Elharit wrote: > > in the CLI i have : > > > > > > 1) for C

Re: [asterisk-users] asterisk and IVR

exten => 534,n,NoOp(Dial status is ${DIALSTATUS}) exten => 534,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]) exten => 534,n,Goto(home,s,1) how to do in order to go to the next if the result is answered exten => 534,1,Dial(SIP/228, 10) exten => 534,n,NoOp(Dial sta

Re: [asterisk-users] asterisk and IVR

ok thank you i will verify and i will update you thanks for your help 2013/7/25 A J Stiles > 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(S

Re: [asterisk-users] asterisk and IVR

exten => s,1,NoOp(User 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 > On Thursday 25 July 2013, Salaheddine Elharit wrote: > > i have asterisk 1.4 insta

[asterisk-users] asterisk and IVR

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 => s,n,Background(${sound

Re: [asterisk-users] block certain numbers

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 > On Monday 1

[asterisk-users] meetme configuration

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 passwor

[asterisk-users] sendmail when no response

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

Re: [asterisk-users] how to launch a URl when dialing a number

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 > El 31/05/13 09:21, Salaheddine Elharit escribió: > >

Re: [asterisk-users] how to launch a URl when dialing a number

> ** ** > > ** ** > > ** ** > > -Justin > -- > > *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 U

[asterisk-users] how to launch a URl when dialing a number

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

Re: [asterisk-users] dahdi driver not getting install

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 > I thought he said rhel 6.3 > > Sent from my iPhone > > On 11 May 2013, at 2:48 PM, Asghar Mohammad wrote: > > he is using deb

Re: [asterisk-users] question about CDR

use > 506,1,Dial(SIP/223&SIP/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

Re: [asterisk-users] question about CDR

thanks i verify but i don't understanding if can someone give me an example best regards 2013/5/9 Ishfaq Malik > On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: > > hello list, > > > > > > i need your help about cdr ,i have installed the m

[asterisk-users] question about CDR

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 hist

Re: [asterisk-users] hwo to stok variable wiith menu

#x27;${CALLERID(num)}'\, calldate=now()) exten => s,n,MYSQL(Clear ${resultid}) exten => s,n,MYSQL(Disconnect ${connid}) thanks and regards 2011/12/1 salaheddine elharit > Hi Noll, > > all works perfectly thanks a lot for your help and support i really > appreciate it :) &g

Re: [asterisk-users] WARNING[28151] from CLI

a few times? > these are all questions you should ask yourself 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 f

Re: [asterisk-users] WARNING[28151] from CLI

figure 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 > > >

Re: [asterisk-users] WARNING[28151] from CLI

signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel => 1-15,17-31 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

Re: [asterisk-users] question about zapata.conf

ok thanks for your help and support i really appreciated 2013/3/26 Tzafrir Cohen > On Mon, Mar 25, 2013 at 10:44:47AM +0000, Salaheddine Elharit wrote: > > hello list, > > > > i have a question related to zapata.conf,if i do any change in > zapata.conf > > i m

[asterisk-users] WARNING[28151] from CLI

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 avail

Re: [asterisk-users] question about zapata.conf

our ZAP 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 fro

Re: [asterisk-users] question about zapata.conf

Wieling > 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, Marc

Re: [asterisk-users] question about zapata.conf

pplied 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, >

Re: [asterisk-users] Need help about round-robin

scenario. > > > On Fri, Mar 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 >> >>>

[asterisk-users] question about zapata.conf

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 -- __

Re: [asterisk-users] Need help about round-robin

ween Wimax 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 ha

Re: [asterisk-users] Need help about round-robin

Hello bharat, ok thank you so much for your help and support now i understand :) 2013/3/22 Bharat Lalcheta > 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" > wrote: > >> ok thank y

Re: [asterisk-users] Need help about round-robin

=yes) > exten => > _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 < > sa

Re: [asterisk-users] Need help about round-robin

{EXTEN},30,A(this-call-may-be-monitored-or-recorded) exten => _0612.,n,Hangup(); thanks and regards. 2013/3/21 Bharat Lalcheta > File is ok there is no etc/zapata file. > On Mar 21, 2013 9:42 PM, "Steve Edwards" > wrote: > >> On Thu, 21 Mar 2013, Salaheddine Elha

Re: [asterisk-users] Need help about round-robin

i mean the burden-sharing between Wimax and FH 2013/3/21 Bharat Lalcheta > What do you mean by roundrobin here > On Mar 21, 2013 8:27 PM, "Salaheddine Elharit" > wrote: > >> hello list, >> >> i have installed 2 diguim cards in my server using ast

[asterisk-users] Need help about round-robin

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 pl

Re: [asterisk-users] issue with inbound calls

--- > > *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**

Re: [asterisk-users] dahdi-linux dahdi-tools and libpri/libpri-

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 > > /usr/src/dahdi-linux-2.6.1/drivers/dahdi/xpp/xdefs.h:152: error: > > conflicting types for âboolâ

Re: [asterisk-users] round-robin in asterisk 1.4

thanks leandro how can i use that line in extensions.conf ? 2013/1/29 Leandro Dardini > The simplest way is to use the Random function and to pickup one number > from 1 to 3 and use that line. > > Leandro > > I am typing from my mobile phone... > Il giorno 29/gen/201

[asterisk-users] round-robin in asterisk 1.4

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 : con

Re: [asterisk-users] block one number in incoming calls

> > > 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 . > > > > U

Re: [asterisk-users] block one number in incoming calls

com] *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 > > ** ** &g

Re: [asterisk-users] block one number in incoming calls

ALLERID(num)}" = "0666XX" ]?3:4) exten => 520xx,3,Dial(SIP/224, 30) exten => 520xx,4,hangup 2013/1/14 Salaheddine Elharit > thanks danny > > > > i think i didn’t explain correctly may question > > > > i revive a lot of calls from this num

Re: [asterisk-users] block one number in incoming calls

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 > Exten => _0666XX,1,answer() > > Exten => _0666XX,n,playback(tt-monkeys) > > E

[asterisk-users] block one number in incoming calls

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 hav

Re: [asterisk-users] IVR: Dealing with database and returned variables

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

[asterisk-users] how to get the Record_ID

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

Re: [asterisk-users] hwo to stok variable wiith menu

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

Re: [asterisk-users] hwo to stok variable wiith menu

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

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