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

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

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

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

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

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

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

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

All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi All, I am new to the list and have ordered myself a Asterisk Developer's Kit (TDM). I am just waiting on the order from a reseller of digium. I have yet to play around with the system when it arrives. And I haven't looked into the manual yet. I do hope I have all the hardwares to test first before I go and buy a sipura spa-2000 unit. I have my sister who is living in Germany.

I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server