similar to: What hardware would you recommend?

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT

I've tried to find some details on the wiki, but was unable to get a satisfactory result, so I am asking here: I have a Linux (FC3) box with these specs: vendor_id : AuthenticAMD cpu family : 6 model : 3 model name : AMD Duron(tm) Processor stepping : 1 cpu MHz : 797.388 cache size : 64 KB MEM: currently 256, looking to upgrade to 512/768

There'll be whiners on all lists, and no list is ever going to be pleasing everybody... (And everyone should decide for themselves whether or not they belong to that group, as I will not be mentioning names! I'm sure there are those who'll think *I* am a whiner...) So at some point just let it be and continue the thread. Keeping on and on about it only increases noise to post

Does anybody know whether somebody ever implemented (linux) drivers or a chan component for the Com-On-Air cards, or is the only way to make it work the use of a Windows box as Dect to SIP gateway for *? (Or does it work at all?) I can still get the cards, which would be great to retire the old DECT based PBX, but it would be of little use if it's not going to work, now is it? ;-) --

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

Hi all, I am struggling with the following and I can't get it work: In the Netherlands where I live it is possible to use special codes (aka vertical service codes) to set special 'behaviour' of phonecalls. So e.g. when I want to dial out with a normal phone and I dial *31*<phonenumber to dial> then it will turn off my numberindication (CID) at the called party. They seem to

Hi All, The wiki seems to be struggling this evening. Anyone else seeing this? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

All Any body know of a generic bluetooth adpater for the universal 2.5mm headset jack on a cell phone that supports the wireless profile *NOT* the headset profile I know jabra has the A210 http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/ JabraA210 but it only support the headset profile .. I am trying to shoe horn my current braindead cell into DocknTalk

Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Sorry if this not the right place to post this.... BUT... Since May 31st, ALL of these user list messages have been filtered by "spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as spam. - Have no clue why this is happening. Luckily, "spamassassin" sent the messages to the "probably-spam" folder on the Linux box & I was able to

Folks, I am a newbie to the VOIP world and have a question (might as well sound silly to some). I would like to set up a PC-to-Phone call from my desktop to a regular PSTN number. Does the Asterisk PBX itself act as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider for this? Are there any free IP-PSTN gateways which I can subscribe to? Do I need any specific hardware to set up such

I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:110@siptest.dmclub.net) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

"Unable to login to the MySQL database 'tiki' on 'localhost' as user 'tiki' Go here to begin the installation process, if you haven't done so already. Lost connection to MySQL server during query" Hi, Everyone: What is the URL of the "list archive"? The 'wiki' seems broken. I want to connect to a single home POTS from an internet

I apologise for writing again about the problem with using stepAIC + multinom, but I think the reason why I had it in the first place is perhaps there may be a bug in either stepAIC or multinom. Just to repeat the problem, I have 126 variables and 99 cases. I don't know if the large number of variables could be the problem. Of couse the reason for doing a stepwise method is to reduce this

Hi, Sorry to bother you on this mailing list, however I tried everything else and I am desperate to get this running. Please send me any hints you can think of. I have installed openssh-3.7.1p2 on a ppc target and trying to connect to an sshd running on a redhat 9 with openssh-3.5p1. I keep getting the error "Disconnecting: Corrupted check bytes on input" no matter what I tried. I

Hi- Anyone using the Junghann's quad BRI card and the Digium E100P in the same system? I'm having a configuration problem where I can configure the cards one at a time (with the appropriate drivers loaded) in a system, but when I try them both together, neither will work. They both work fine one at a time. Probably has something to do with the channel numbering. I've tried

Greetings - We've a pair of ISDN BRI that we use for dialtone, fairly happily except for the recent meltdown of one of our Netgear RT338's. We're in the middle of slowly migrating to a VoIP/Asterisk-on-FreeBSD based phone system. I had originally considered just buying a Digium TDM400 card and continuing to use the RT338's to bring out POTS lines from the BRI's, but the

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part