Displaying 20 results from an estimated 6000 matches similar to: "Delete voicemail"
2005 Jul 01
19
Epia C3 Linux
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
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2007 Apr 12
6
Fax Blast over IP?
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>
2005 Jun 07
3
AAH 1.1 - CRM Setup
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
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2005 Mar 11
7
Sip show registry returning nothing
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
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2005 Mar 09
6
VoIPJet
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
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2005 Feb 15
14
X-Lite Softphone
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
2005 May 11
5
IAX.CC/SixTel
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
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2005 Mar 11
8
No ringback over IAX - LiveVoip
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
2005 Oct 06
3
WCFXO and T1 PRI Card?
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
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2005 Mar 10
7
IAX2 800 Termination
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
2005 Jun 02
3
asterisk on internet sip phone behind nat - doessomeone even have this working
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
2005 Mar 04
1
defold usernames in asterisk@home version 6
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
2005 Aug 09
3
First PRI
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
2005 Jun 13
7
MCI vs. XO/Allegiance
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
2005 Mar 29
1
External voice channels pack up
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000066">
<small><font face="Verdana" size="-1"><small><big>Hi
2005 May 27
1
VoiPSupply Dot Com: Epilogue
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
2005 Feb 28
2
Fax Failing
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
2005 Mar 16
3
(Yet another) Music on hold problemand another...
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
2005 Jun 03
6
Livevoip 800 Choppy Audio
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
2005 Feb 25
3
Festival - Asterisk@home
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,