search for: khalidtouati

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

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>

Hi Guys, i am using the following config in pbx1: register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176> [pbx1] type=friend

Hi Guys, i've connecting two pbx server successfully for several times using the following config: register => USPBX:mypass at 122.11.176.35 <USPBX%3Amypass at 122.11.176.35> [PBX1] type=friend host=122.11.176.35 trunk=yes sercret=mypass context=external deny=0.0.0.0/0.0.0.0 permit=122.11.176.35/255.255.255.240 insecure=very allow=all nat=yes qualify=yes canreinvite=no in the other

Hi Guys, i could turn debug on in a asterisk 1.6 box (by enabling debug in logger.conf and core set debug to > 0), but my issue is i cannot enable debugging in a 1.2 box by doing the same 2 steps, also this is a production server so i can't restart with debug enabled, do you guys know how i can turn debug on or just know why it's not getting enabled? Thanks a lot for your help! --

Hi All, i have a question, is there any soft-phone available for Black Berry use, I've been told there is a firefly one, but when i looked, i found nothing, is any body has an update on this please? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/1bf1a72e/attachment.htm

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

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats

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 Guys, i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1) and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue i'm having is that i'm able to receive faxes from a website (that offer this service) but not able to receive from a regular fax machine (that is working perfect). [fax-rx] exten => receive,1,NoOp(**** FAX RECEIVE ****) exten

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

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

Hi Guys, I'm having a non-obvious issue, i am using Fax for asterisk to receive faxes, so when i test using a website that send faxes it's working great: the fax is received and the fax2mail app is called and i get it in my email box. but when i try using a regular fax machine everything in logs (turned on debug) but all of the sudden a line appear saying: Channel 0/1, span 1 got hangup

Hi All, I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan: exten => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr) exten => 8355,n,Dial(IAX2/8366,48,tTWwr) (i made that simple to exhibit issue) I got just 1 ring in 8366 extension before it hangup, what i noticed is the total time spent on ringing is 30s that means if i use 12s in the first dial i get 18s left

Hi Guys, Do you know of any asterisk community version that does video 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

Hi Guys, i have recently connected my (working) asterisk 1.2 server, with two 1.4 asterisk servers (one using SIP the other using IAX), since then (i believe) people starts complaining about a high background noise when using the handset on Polycom phones (but when using the speaker it's fine, and i noticed that my self), my question is, can anybody tell me any step to begin diagnosing the

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

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

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