search for: walid

...I am not that expert so excuse my ignorance with this subject. So please if anyone has any useful information or is sure that this can now work please send me whatever you have on that. I simply want Asterisk users to get their dial tones through CCM. Thanks and I appreciate your assistance. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050625/84ccdfe4/attachment.htm

...a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either. Your input is highly appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050109/cd73d10a/attachment.htm

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

...t 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 was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041204/1cd2c2a2/attachment.htm

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

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050112/0376094c/attachment.htm

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

...owulf mailing list, and did not get any responses, as i have tried different Centos kernels to see if the behaviour changes or not, and it did not change much, I am posting it here, I hope no one minds, and thanks in advance for any pointers or clues? ---------- Forwarded message ---------- From: Walid <walid.shaari at gmail.com> Date: 2008/6/12 Subject: RHEL5 network throughput/scalability To: beowulf at beowulf.org Hi All, I have an issue with a new cluster setup where the nodes are RHEL5.1(with the latest 5.2 kernel), when i try to write NFS data, the nodes scale linearly until they r...

...release? Is there anything equivalent for CentOS, or is this RedHat only behavior? TIA, -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid Jumblatt

I have installed Asterisk@Home on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/99d6d45f/attachment.htm

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

...through 10 as an x-intercept. For the first case i have used the (y~x-1) syntax, but the linear model returned a regresssion that didn't pass trough the points (with a higher Rsq than expected). For the second case, i did not find a solution yet.. Could somone help me please? Thanks in avance Walid SADOK

...ot an rpm/yum heavyweight. I'm sure, however, others may appreciate the answer. TIA, -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid Jumblatt

...case 1 as well. No big deal. I'm just wondering if anyone can explain this behavior. -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid Jumblatt

...seen this type of problem with a touchpad, and does anyone have knowledge of a fix? TIA, -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid Jumblatt

...d (Centos4), and it works just fine. Just wondering why it's not in the dag el4 repo? -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid Jumblatt

...h shows our problem: PC1 -> SRW2024 <- PC2 PC1(192.168.1.1) was booted with knoppix, terminal server enabled PC2 was set to PXE network boot in bios screen of PC2 : --- >>> --- >>> --- >>> --- NVIDIA Boot Agent 220.0518 [...] CLIENT IP: ___.___.___.___ <- got walid ip from PC1 dhcp server GATEWAY IP: ___.___.___.___ <- got walid ip from PC1 dhcp server PXE-E11: ARP timeout PXE-E11: APR timeout PXE-E38: TFTP cannot open connection Reboot and select proper Boot device --- <<< --- <<< --- <<< --- Any other switch works fine with t...

...ny 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. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041222/dcf75111/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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/21f88451/attachment.htm

Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have => 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXXXXXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XXXXXX instead of