search for: shoval

...redhat 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. We are only familiar with red hat and are in a point in time that switching distros is not available. The guy installing the system is already on location. Yes, I know we made a silly mistake. Please help us... Thanks. Shoval

How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm

...'t figure how to configure the thing. I know I'll need to configure the regular PBX to forward certain calls to the lines connected to asterisk (by prefix, or just have everyone dial 8 and get a line) Does this scenario make sense to anyone? Or am I barking up the wrong tree? Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031105/34bb994e/attachment.htm

Hi all. Can someone recommend a good UPS for using with an * machine that provides some linux tested software to do managed shutdown in case of power loss? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever

...to make calls from * extensions to Panasonic extensions and the other way around. Making outgoing calls from extensions one one side to lines on the other would be nice too. I can put another * machine at the main office, but what is the best way to integrate it with the Panasonic PBX? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas?

...Caller id, internally, works just fine (as long as I use numeric only callerids) for IAX and grandstream. Is there a way to have the analog phones' LCD display show the caller id? These are plain old regular analog phone, that if I had callerid from my telco would show on the screen. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

...than for self use, but mostly the same. I've seen posts where real-life installations are mentioned, but not a reference to how Asterisk is working on production (and productive) environments. Any experiences would be very welcome I believe, not only on pure technical, but wider, sense. Thanks Shoval for raising this issue, and to All for keeping this list so alive. Best Regards Jos? L. Perez ----- Original Message ----- From: Shoval Tomer To: asterisk-users@lists.digium.com Sent: Sunday, November 02, 2003 16:36 Subject: [Asterisk-Users] a bit frightened, guys Hi, I started looking...

Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. N...

...;sip for iconnect type=friend username=12345678 secret=1234 host=natrelay.deltathree.com [1000] ;my sip extension type=friend secret=1234 iauth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=rfc2883 callerid="Tomer Shoval" <1000> disallow=all allow=gsm context=default Extensions.conf [general] static=yes writeprotect=yes [globals] [default] exten => _91.,1,Dial(SIP/${EXTEN:1}@iconnect,70) exten => _91.,2,Hangup As you can see, very minimal, for this kind of usage (outbo...

Hi, Where can I find a complete * manager api guide, the one one wiki is missing informations like the monitor function for example, Thnx Serge

...For instance, can I get the hardware and return it after a while if it's not working? Where? This way, if it's a no go, I'm not stuck with useless hardware that cost thousands of dollars. In short, can Asterisk be deployed to a real production environment, like ours? Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/42c0b361/attachment.htm

...ms ok... I tried changing my sip conf to include allow=g729,gsm Now the calls sounds fine, but the bandwidth is uses is near 20K instead of just 6K (both phones are near me, and the Asterisk server is at a remote location, and I can monitor bandwidth usage in my FW). Can anyone help? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says "Premature end of script headers: vmail.cgi" Shoval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031030/397453ce/attachment.htm

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

...gateway (Alejandro Ruiz) 3. Re: asterisk bandwidth management (Andrew Kohlsmith) 4. Re: Best or any VoIP provider that works with *? (Andrew Kohlsmith) 5. Re: ISDN PBX + IVR + Voicemail Configuration - Sanity Check ... (Klaus-Peter Junghanns) 6. RE: archives gsm of asterisk ??? (Shoval Tom) 7. Re: Red Alarm (Andrew Kohlsmith) 8. Re: IAX/SIP Client (Dan) 9. Re: USB handsets/headsets?? (Dan) 10. Re: Anyone using * in a live production environment? (Andrew Kohlsmith) 11. Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...) (Chris Hi...

Rich and I have updated the Wiki page "Asterisk rollout tips" with advice on how to plan and implement your Asterisk rollout. This page is based on many discussions on the mailing list, so don't be surprised if your comment or thought is included in the text. Thank you for your input! http://www.voip-info.org/tiki-index.php?page=Asterisk+rollout+tips Any comments are welcome! /O

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

...boards, a pci display card, and a network interface. And I know I've had machines at home that had a display adaptor, modem, network, scsi, and soundblaster all together. Yet, people claim it won't work because of lack of IRQs, and that it's not related to Digium. What am I missing? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200