similar to: small business installation.

Hi, We've got IAX softphones, GrandStream VOIP phones and zaptel connected analog phones. 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

Hi, Anybody have an idea how to integrate * with a Panasonic td-1232? We one at the main office, and are installing * in a branch office. We'd like to be able 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

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

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. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? c. where can I find help for configuring it? d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd.

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

Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX - is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

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

Hi. Please excuse me asking this again. But it really puzzles me. We're installing asterisk at a branch office at NJ (HQ is at Petach-Tikva) It'll need to support 5 POTS lines, 11 analog extensions and four VOIP phones. I wanted to go with a T1 card from digium and a channel bank, but we have a dead line. It has to be up and running by January 1st. I don't have the time to start

Hi, I believe the issues raised by this message are the same as mine, more on a commercial sense 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

Hi, I started looking into asterisk cause we're looking for a real-world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization - to save on long distance and international calls. I've been

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hi, I was able to connect asterisk to iconnect's service. It took me almost two hours, but it's because I was having NAT trouble. I finally discovered that you can set the iconnect host to natrealy.deltathree.com to make it work. (for those of you who, like me, don't have the time to search the archive I'll provide a working sample in a minute) My problem was sound

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. 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

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