similar to: Delete voicemail

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 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>

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

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

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

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, 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

Can I have a TDM400 and a T100P in the same machine? I am using AAH and trying to combine two boxes. If so, can anyone tell me the proper config for zaptel.conf and zapata.conf? Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051006/061d9d84/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

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

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

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, 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

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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

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

Hello All, I installed Asterisk@home with no problems whatsoever. All features so far work great. However, I have been trying to setup the festivval weather AGI script and it won't work. I see the script fire off in the CLI and it completes with no errors. However, I never hear anything on the extension. Does anyone know if there is something undocumented that I should have done? Thanks,