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...he 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

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

...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

...ks 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

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

...nsions 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

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

Hi all. Can anyone comment why shouldn't we use FC 3 for an * production system? I'm not looking to start a distro war, but we just found out that 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

...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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 -------------- next part -------------- An HTML attac...

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 a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

...l that extension, the first machine will answer and ask for the extension number of the other end. So what you actually have, is a local extension tha works as if you picked up an extension on the other end. I hope this help... ----- Original Message ----- From: "Shoval Tom" <shoval@softov.co.il> To: <asterisk-users@lists.digium.com> Sent: Wednesday, November 05, 2003 6:12 PM Subject: RE: [Asterisk-Users] Using Asterisk as a VOIP gateway > How is it not economical? > I already have the PBXs on both sides. > If I switch to * I'll need to get a channel bank >...

...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

...bably be much easier for us to set up. > > I can understand and respect telco guys, who've setup channel banks for > TDM PBXs for years not being afraid to try and hook it up with *. > > Me? I'm scared as hell > > > > > Shoval Tomer, > IT Manager, > SofTov Advanced Systems, Ltd. > Office: +972-3-9230686 ext. 179 > Fax: +972-3-9216642 > Mobile: +972-54-8000200 > >

..., 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

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?

...question: I'd like to connect a modem to one of the fxs ports. Can it be used to make dial-up connections via asterisk? (modem > Asterisk > PSTN) Did someone get the FAX extension working (routing a fax call to a fax machine connected to an FXS port?) Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

...ow=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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041220/91386b49/attachment.htm

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried