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Hi, I was just wondering if anyone could help me with this weird problem. I am trying to install an Arabic program using wine. In order to do so, I run: LANG=ar_SA.utf-8 wine install.exe The title in the title bar comes out correctly, however, whatever content the program might have comes out blank, nothing. I just could not figure out why it came out right in the title bar, but nothing comes out

...host=dynamic context=fullaccess My extension.conf is as follows for the server that is behind NAT: [fullaccess] exten => _6XXX,1,Dial(IAX2/2000@66.151.89.145/${EXTEN}) exten => _6XXX,2,Hangup exten => _6XXX,102,Hangup --------------------------------------------- Free POP3 Email from www.Gawab.com Sign up NOW and get your account @gawab.com!!

Hi David, I have a copy of Windows that I bought a year or two ago, but never really used it, as I preferred to use Linux. Got it just in case I needed the license, I guess now I can put it to use. Me paying for the Windows copy should include the fonts, right? It's working now, and showing the fonts. Thanks a lot :-D Now all that I have left is to get the characters to attach, and I

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

...bar=11) # [1] 77 ## So this works, and all is well, nay? Nay. Turn thou to (1) to see what ## doth happen when we call timesDefineInside from inside another function. -- Sietse Brouwer -- sbbrouwer at gmail.com -- +31 6 13456848 Wildekamp 32 -- 6721 JD Bennekom -- the Netherlands MSN: sietse at gawab.com -- ICQ: 341232104

Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid -------------- next part -------------- An HTML attachment

Dear list, I am running Asterisk CVS-v1-0-12 what is this called in terms of Asterisk versions convention? Is it Stable , Head, latest release !!! Excuse me if the question is too basic, but your help is appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head I am a newbie and about to install Asterisk on SUSE Server. Can someone please advise what is the best version type and number should I use. My environment is not so big. I only wish to eventually get my asterisk to talk to Cisco CCM 3.3.4.

Hi guys , i am starting to "play" with qos in linux. Well , i am trying to setup an ingress filter but i do not know why it is not working. tc add qdisc dev eth0 ingress tc filter add dev eth0 parent ffff: protocol ip prio 1 handle 1 fw police rate 160kbit burst 256kbit drop flowid :1 After that : iptables -A PREROUTING -t mangle --sport 80 -j MARK --set-mark 1 So , i think this

[I sent this earlier but I guess the list is subscriber-only?] I just set up wondershaper, it has a simple filter on the downstream direction to limit the bandwidth usage: tc qdisc add dev $DEV handle ffff: ingress tc filter add dev $DEV parent ffff: protocol ip prio 50 u32 match ip src \ 0.0.0.0/0 police rate ${DOWNLINK}kbit burst 10k drop flowid :1 This is effective but is there any way to

Hi All, We have just installed Asterisk@Home. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated.

Hello Every one I need to enable Asterisk to route external land line calls to the PSTN. Regarding our environment, we have Cisco CallManager (3.3.4) to which IP phones are connected. E1 terminated on a couple of As 5300's which are controlled by a soft switch (Cisco PGW200 Call Control). What is the best scenario to route external calls to PSTN. Should I use SIP or just connect Asterisk

Hello, I am having a couple of problems. Any help is appreciated. 1. The voice mail messages arrive in the mailboxes but when I play them back, the IVR tells the time and date of the message but never plays it. It is as if it skips it. 2. Asterisk never seems to send the voice mail as an attachment!! Please see below my config files: Thanks Walid --------------------

Hi, I just want to know what is the easiest way to have Asterisk route calls to PSTN. Hope any one can help me. PS: Any solution using a Cisco device is preferable. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050112/1604835f/attachment.htm

I need to install Oh323 in order to get Asterisk connect to Cisco CCM 3.3.4. I cannot find the dependencies needed for that. Can any one send me link: The readme file says they are at the following links but the exact version isn't. Please help. Walid o PWlib (Portable Text and GUI C/C++ Class Library) download from http://sourceforge.net/projects/openh323 (v1.6.6) (required)

I am trying to build PWLIB to get OH323 up and running. I am not an expert in linux but can someone help telling me how I can do the following: How can I add a directory to LD_LIBRARY_PATH?! Thanks in advance ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- ---------------------- For

Hi, well, I really need your help here. I have tried compiling oh323 many times and I always get the following error when trying to "make opt" open h323. Any ideas?! Compilation Error: -------------------------- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat

I am trying to compile oh323 and having the following error. Can anyone help please?! This is my third post. These are the versions I am using: Compilation Error: -------------------------- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat -lresolv -ldl

Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050116/f75bd2fe/attachment.htm

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: