similar to: IP Intercom's

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I

Everyone-- Here's a cost analysis, rather crude and inspecific, of using Asterisk to implement a phone system. I'm really quite naive and new to all this, so I'd appreciate any corrections, tips, pointers, etc, from those in the community, who have far greater experience, knowledge, skill, etc. than I. Am I forgetting something important? Am I way off in my estimates? The Minimum

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike

Hey guys, I have just developed this problem with my Windows XP box. I think it started since I installed XP SP2. Both SJPhone and Xlite does some kind of bridging with the speaker out port. When ever I make a sip call to where ever, the other party hears a lot of echoing. Well I noticed just now when I was playing mp3's via Winamp, the music was being played through my sip calls that I

On my default asterisk installation, *80 didn't work until I modified the source to move call pickup to *9. I wasn't sure what I was doing but *80 works now. Except I thought *80 would play some voice prompts that gave the option to add the last caller to the black list as well as other options. Instead I just get a studer dial tone after the last caller gets added to the database. When I

Hi All... I'm trying to get the intercom feature working on some snom 190 phones but having no luck... As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. I've email'd snom a few days ago but have yet to get a response. On my 190s, im running snom190-SIP 3.57v. I am pulling the config for the

Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has