search for: wiley

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050701/58373c94/attachment.htm

...stem. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it installed as is? I really do not want to change my version of Asterisk since it is running well at this point. Is it possible to just update and add the card? Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/30c4f494/attachment.htm

...When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Best BootRom & SIP Code for Poly600? (Eric Wieling aka ManxPower) 2. RE: Epia C3 Linux (Michael Stahl) 3. RE: Epia C3 Linux (Wiley Siler) 4. Users handbook (Chris Mason) 5. Re: HW Capacity plan - Asterisk capacity per Xeon 3.0GHz (Manuel Soto) 6. Re: Epia C3 Linux (JD Austin) 7. Re: Remote SIP Connections (Blake Krone) 8. RE: Epia C3 Linux (Rob Thomas) ----------------------------------------------------...

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

...e 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/224c3e2a/attachment.htm

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050613/cd5fc25b/attachment.htm

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> www.education2020.com <http://www.education2020.com/> ----...

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

Dear group, Here is my environment: > ls() [1] "DailyPL100419" "DailyPL100420" "DailyPL100421" "dd" "i" "l" "PLglobal" "Pos100416" "Pos100419" "Pos100420" "Pos100421" "position" [13] "result" "sel"

...r would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can anyone with experience in this department let me know if they think this idea is possible? Thanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities oth...

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

...g. Just a fast busy. Outgoing works perfectly however. I have a defined DID in the AMP interface and verified it is written to confs and have reloaded. Can anyone tell me another way to verify that something is coming in? Or did I just miss something on the whole IAX over NAT? Thanks all, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/68f748a2/attachment.htm

...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 part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040814/c8f6aafe/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

...PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I will check out the queue system you referenced. Thanks! Wiley -----Original Message----- From: Seth Remington [mailto:sremington@saberlogic.com] Sent: Saturday, July 17, 2004 7:45 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Using a group variable for a groupofextension to dial It would ring all three at the same time. You are pr...

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

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

...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, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/7e882947/attachment.htm

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

...be used in regular script files. When I am still working on a script, I often want to being using what is done, but I would like the parts I am still working on not to be run when I use source() on the file. I can set everything off with #, but that gets to be tedious. Thanks, Josh -- Joshua Wiley Ph.D. Student, Health Psychology University of California, Los Angeles http://www.joshuawiley.com/