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Hi Guys/Gals - I don't post here often but I read with interest all the postings. - I've been on a lot of mailing lists, but this one is by far the most interesting. I've been doing a lot of work with 'tftp' loading Cisco 79xx phones with firmware, configs. for asterisk, etc. And I see where a lot of folks have trouble with 'tftp', use alternate port numbers

Hello all - This is basically an updated re-posting of one I've posted a few days ago. Thanks to the kind help provided but I still can't make it work. But I'm moving a little further down the line (thanks to you folks). Basically, I've got an Asterisk server in a LAB ENVIRONMENT on my home LAN. The server has a Wildcard TDM400 installed but has no POTs lines/phones connected.

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

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:

Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically

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

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

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

Hi Folks - This may sound weird - but here goes: I live in Japan and on my home POTS line I have a Fax/Phone machine. If I receive a fax, the thing automatically switches to 'fax mode' and prints the fax. If the call is a 'voice call', it sits there & rings until answered. The above is very reliable and works okay. Of course signalling differs in each country (and even by

Sorry to clutter up the mailiing list, but I've been unable to post to this list for the past 2 WEEKS! My ISP's blocking SMPT from other than his own servers. I think I've worked around it. - But if I see this message in the digest then I know I'm okay. Again. - Sorry for any inconvenience. Gary Guthary

Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... -------------- next part -------------- A non-text attachment was scrubbed... Name: rhuddleston.vcf Type: application/octet-stream Size: 575 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/bbb0a930/rhuddleston.obj

I've been reading this forum for over the past 4 years and have gained a wealth of knowledge. - Thanks to all! I don't post very often but I've just ran into a problem/condition that I simply can't figure out. - Hopefully some kind soul will help me. I've got an Asterisk server in a lab environment on my home LAN. - The LAN is "talking" to the Internet via a Linksys

I'm trying to get shared folders to work, and I've noticed a few issues. I created a private and public namespace, the private one with autodetection for Maildir or Mbox, and the public one for a public folder. Something like this: namespace private { separator = / inbox = yes prefix = location = } namespace public { separator = / prefix = Public/ location =