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...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> www.education2020.com <http://www.education2020.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070412/e30d3fb0/attachment.htm

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:

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/> Helping students on a mission. Graduation and beyond. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/a...

...but may not support all of the instructions (causing a crash). The via arena site gives instructions on how to compile and get it installed on your processor! (I have the C3 Nehemiah processor so I didn't need to recompile) ________________________________ From: Wiley Siler [mailto:wsiler@education2020.com] Sent: Tuesday, July 05, 2005 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All...

...to (ext-fax,in_fax,2) -- Executing Macro("Zap/2-1", "faxreceive") in new stack -- Executing SetVar("Zap/2-1", "FAXFILE=/var/spool/asterisk/fax/1109597736.32.tif") in new stack -- Executing SetVar("Zap/2-1", "EMAILADDR=fax-recipients@education2020.com") in new stack -- Executing RxFAX("Zap/2-1", "/var/spool/asterisk/fax/1109597736.32.tif") in new stack -- Executing Hangup("Zap/2-1", "") in new stack == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/2-1' -- Hungup ...

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, 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

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

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

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

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

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 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

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

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

...and OSX. I learned routing and networking as needed for the job and look at this project as just more learning. And I'm VERY excited that I found an active community to query. Thanks again I hope that helps tell the story and background a bit more. -brandt On 5/27/05, Wiley Siler <wsiler@education2020.com> wrote: > OK. Here are some pointers.... > > Be sure to read up here... > www.voip-info.org > > Lots of info and very specific to SIP, Asterisk, etc, etc... > Also, with SIP, firewalls become an issue. Check out the SIP stuff at > above URL. > > Next, you will...

Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com

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

No argument here..... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Esben Stien Sent: Thursday, August 11, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation "Jonathan k. Creasy"