similar to: Voicemail Size on Disk

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

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

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

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

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

First, I am surprised at IAX... My asterisk server is behind a firewall, I am behind a firewall, and my iax client connected... Voodoo??? :) But, the firewall has one unblocked port, which I think I should set to the IAX, so the magic won't be necessary, and it makes sense to think that there will be less latency without the "magic layer". But, when I set port=myport in the

Hi, How do I put a call on hold? If i press # the music on hold plays to the other person, but asterisk asks for a number to transfer... I don't want to transfer, I simply want to put the person on hold, so he/she can hear the music while I do something, then get them off hold. Is it possible? The scenario: The person calls me from a SIP phone, and I receive the call in a regular PSTN phone,

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

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

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

Hello list, I've posted my problem on BSD list and i still have the problem. The remote side receives the call , but there's no voice on the call. I tried everything about possible NAT problems .. but ther're on same net. My platform: FreeBSD 5.2.1-Release Asterisk 1.0-RC2 soft phones : X-Lite >>>> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20")

I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason

Hi there, I''ve read the paper on Xen live migration, and it shows some very impressive figures, like 165ms downtime on a running web server, and 50ms for a quake3 server. I installed CentOS 5 on 2 servers, each with 2x Xeon E5335 (quad-core), 2x Intel 80003ES2LAN Gb NICs. Then I installed 2 DomUs, also with CentOS 5. One NIC is connected to the LAN (on the same switch and VLAN), the

Hi there, I was looking for a GSM gateway, but I didn't find any prices... Anybody knows how much one of these costs? And, is it possible to use an amateur radio with asterisk? Thanks, Marconi.

Hi there, I got really tired of false hangups, specially when someone calls from SIP/IAX/whatever to PSTN, which makes no sense in using busydetect in the zap channel, as the caller will eventually hear the busy tones and hangup, causing the zap channel to be "freed" as well... After suggesting that the busy detect be enabled only for inbound calls, and finding other people that think

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server,