Displaying 20 results from an estimated 103 matches for "wsiler".
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siler
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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2005 Jul 05
1
Help with Cisco 7905G corrupted image!!
...staller 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...
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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2007 Apr 12
6
Fax Blast over IP?
...e 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/>
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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 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 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 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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2007 May 14
0
Asterisk Now
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.
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2004 Aug 14
7
Free MOH MP3
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
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2005 May 10
2
Manoj Shetty is out of the office. [Email checked- EMEA]
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
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 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
2004 Aug 10
3
Polycom IP 500 - MWI Not Working
Hello All,
I have Polycom IP 500 phones which I would like to have message waiting
indicators on. So far, I have my system running well but the problem I
am seeing is that MWI doesn't seem to tell my phone that it should
display a MWI state. The light does not show when you have message nor
is there any indicator on the text lines of a message waiting. The wiki
doesn't cover this
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 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
2004 Oct 06
10
Asterisk and SIP phones
I have Asterisk server providing phone service for my company.
The server is behind a PIX-515 FW and is assigned a private address
192.168.11.X/24.
With that said what is best to provide remote SIP phones (home offices)
securely.
If the solution is to put up another Asterisk server with a public IP
address I am opposed to that.
I am looking for the a secure reliable solution to set up remote SIP
2004 Jul 19
0
Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
...ter interface and if you ever
> press the eject on the IPK you could use the t-1 as a PSTN interface.
>
>
> --__--__--
>
> Message: 3
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
> Date: Mon, 19 Jul 2004 16:28:25 -0700
> From: "Wiley E. Siler" <wsiler@e2020inc.com>
> To: <asterisk-users@lists.digium.com>
> Reply-To: asterisk-users@lists.digium.com
>
> Mine does the same. Once in Message center I can choose selection
> 1.Message Center and then soft key Select. Then I select the
> registered line that I want to che...
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 May 27
1
Fwd: Newbie here. Tips on setting up 100 phones wanted.
...indows, 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.
>
>...