similar to: Help with Cisco 7905G corrupted image!!

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 AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a "normal" hdd. Can that cause any problems? BR Amund Nygaard -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til:

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 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>

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/>

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

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

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

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

dude it's gotta be something with your system. Im using same setup at home with a TDM22 with no probs. Asterisk@Home 1.5, Compaq Deskpro EN P3 500, cordless phones. You did a ton more than I did, I basically plugged everything in and installed a@h.Worked first try. Hate to say it, but would it be possible to try a different system? Even an old system as long as it is PCI 2.2. The big clue

Hello All, My system is built on a dual Athlon box using Redhat 9. From time to time there are problems with the phone system that are hard top track down and sseem to be resolved with just a simple reboot. I read somewhere that Linux loves teh Intel platform and that on AMD it is not as good. Is this true or just hooey? If so, would I get more reliability by dumping the AMD box in favor of

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

OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to

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

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

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

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

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

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

>From atacomm.... ________________________________ From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Thursday, May 12, 2005 2:24 PM To: Wiley Siler Subject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation