search for: touati

Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a

Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the

...the following in pbx1: -- Executing [18021 at default:1] Dial("SIP/8029-b7413678", "SIP/pbx2/8021||TWw") in new stack -- Called pbx2/8021 [Mar 4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite: Received response: "Forbidden" from '"Khalid Touati" < sip:8029 at 172.16.200.176 <sip%3A8029 at 172.16.200.176>>;tag=as1dcf5ff2' -- SIP/pbx2-09cf4468 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8029-b7413678' status is 'CONGESTION' though i am u...

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hi, I've been browsing for FXO devices, and I'm really surprised at their costs. Why such devices are so expensive and somehow hard to get ? Samy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/a92ee327/attachment.htm

Hi Guys, do any body know how to receive debug info on RxFAX application? i am experiencing a lot of fax failures and can't guess the reason behind. Thank you very much for any help! -- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100402/cf26573a/attachment.htm

...ideo codec trans-coding or in other words supports video? I have 1.8.8.1 and I see h263.c format files but can't see that codec in make menuselect. it might be just a license issue (if h263 has to have license), but not sure if community versions offer video calls at all. Thank you! -- Khalid Touati -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140901/4f4f5b16/attachment.html>

Hi List, I did follow the procedure to install Free Fax for Asterisk successfully till i came accross this isssue: i can't load the fax module: pbx3*CLI> module load res_fax_digium.so Unable to load module res_fax_digium.so Command 'module load res_fax_digium.so' failed. [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module

Bonjour, 2008/2/22, Sid Touati <Sid.Touati at inria.fr>: > Bonjour > > > - fichier config.log, dans le r?pertoire de PSP, apr?s avoir lanc? "./configure" > > > le fichier est joint > > > > - r?sultat de "rpm -ql lib64sigc++2.0-devel" > > # rpm -ql lib64sigc++2....

Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 20000, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as

_____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, April 13, 2010 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Time variables in system application Hi Guys, i have a weird thing here: when using time variables (%F & %T) in a shell script, out of dial plan (particularly system()...

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC

...learning LLVM from. linearscan gives the best results among LLVM's allocators so it also should be the first allocator to compete with :) If you think about better register allocator I'd suggest you to look at http://citeseer.ist.psu.edu/kong98precise.html and find a paper of Sid-Ahmed-Ali Touati named "Register Saturation in Instruction Level Parallelism". These works propose faster and better algorithms than graph coloring, at least as I understand it. Best regards, Anton. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/...

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

Toward a better register allocator, I'm attempting to understand the dataflow information available to the allocator. What's the difference between LiveInterval information and LiveVariable information? If a LiveInterval is based on a linear ordering of the machine instructions, isn't it rather conservative in nature? Let's say I have a typical diamond CFG: A

Hi all, I am running to the following problem, when using the below dialplan to receive fax, everything works perfect till this line exten => receive,n,ReceiveFAX(${FAXFILE}): and then the following line cannot be executed, it's like asterisk can't go back to dialplan and continue, the good news is when i check what is received in my fax folder i find that the file is a valid one (not

Hello, I am new in this mailing,please if somebody knows how to make kdump able to load in startup. i will be very thankful, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20090409/d3435a68/attachment-0002.html>

I am trying to view the source code of the function nlm in the stats package of R 2.4.1. I downloaded the source from CRAN and opened nlm.R, and it calls a .Internal function: .Internal(nlm(function(x) f(x, ...), p, hessian, typsize, fscale, msg, ndigit, gradtol, stepmax, steptol, iterlim)) This is the same thing I saw when entering the function name at the R command

Hi, I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance. I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1 version. Will the I2 version work in Canada with regular anlog phones, or will I need to change it. Thanks for your answer. Samy -------------- next part -------------- An HTML attachment was scrubbed... URL: