similar to: Fax Blast over IP?

Can someone tell me what is included in this distro? Does it have voicemail, meetme, panel, and IVR? Thanks, Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com> www.education2020.com <http://www.education2020.com/>

Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a

Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050607/b18b3743/attachment.htm

Anyone have an opinion about these guys and their recent performance? I need some local DIDs and they provide for my area code.... Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050511/365bc7b0/attachment.htm

LOL - You mean he actually 'met' Newt Gingrich? How dare you not extend him credit!!! I mean seriously... For such a distinguished individual... Hey, not only have I met the heads of several multi-billion dollar corps, I have gotten absolutely blasted drunk with them. So I should get credit, a 40% discount, and your daughters phone number, right??? LOL Seriously, though. I think it

I am looking for a good provider for IAX2/800 termination. I am currently using FreeWorldTel and wanted to use NuFone but it seems that both of them don't provide customer service. FreeWorld has terrible voice quality and NuFone never answers their phone or responds to messages. Thanks, Linn

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be

So in order to answer the background and backbone questions here is the system as it is. I hope this isn't too much for the list but I'll post it in response to a few inquiries. The current system is quite interesting. We have an office in a town that is about 50 miles from the ski area. The ski area is powered 100% of of generators and the telephone access and internet access goes from

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

Whew... What a relief. I know the list was worried about why we could not get a hold of Manoj Shetty.... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Manoj Shetty Sent: Monday, May 09, 2005 12:24 PM To: asterisk-users Subject: [Asterisk-Users] Manoj Shetty is out of the office. [Email checked- EMEA] I

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default