Disclaimer at the bottom still looks ridiculous even in Spanish... LOL
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:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
www.education2020.com
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:[EMAIL PROTECTED]
Can anyone tell me which config file tells the phone what file to load
as bootrom.ld?
Or is this hardcoded in the phone? I just got a IP501 but I have a
bunch of IP500s...
Will the bootrom (2.6.2) work OK with both the IP500 and 501?
Thanks!
Wiley E. Siler
Director of Information
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
, April 12, 2007 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Blast over IP?
On Thu, 12 Apr 2007, Wiley Siler said something to this effect:
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP
?
Wiley Siler wrote:
Basically, I want to send bulk faxes to a list of my clients.
It is time consuming for a person to individually fax so a blast type
solution seems best.
Over IP is of course to save money...
Thanks for the link, reading now...
Any suggestions for the blast then?
My
Are these guys still around? I cannot get to www.nufone.net or
nufone.com
Thanks,
Wiley
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Nufone
Wiley Siler wrote:
Are these guys still around? I cannot get to _www.nufone.net_
file://www.nufone.net or nufone.com
Not only can I get to their website, but yesterday I called their
customer service and for the first time ever
Hmm... Wouldn't you just place something in t,1,
To catch the timeout event and loop back to the top of the IVR?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert P.
McKenzie
Sent: Thursday, March 09, 2006 2:21 PM
To: Asterisk Users Mailing List
Hmm... And Nufone is down suddenly Coincidence or other?
Stated reason was multiple hardware failure.
Somehow I am betting anyone with this problem already noticed too...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Development
Excellent to know. Fortunately for me I don't have any scheduled use on
my DID from them today.
Phew...
I am surprised that hot swaps are not more common practice.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday,
.-Rusty
On 1/25/06, Wiley
Siler [EMAIL PROTECTED]
wrote:
Hmm...
And Nufone is down suddenlyCoincidence or other?Stated
reason was multiple hardware failure. Somehow I am betting anyone with
this problem already noticed too...W-Original
Message-From: [EMAIL PROTECTED
I am going to assume the best and hope it was a an issue of testing code
missed at release.
How or why that would be the case is beyond me but I sure hope that is
the case.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical
Support
Sent:
All my Polycoms are set to...
dtmfmode=rfc2833
Should solve your problems.
Best configuration is through the config files and using an FTP or TFTP
server.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: Wednesday, January 18,
VoipJet has been great to me for dial time.
Nufone.net is where I get my inward dialing for my VoIP. Also good
experience so far.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, January 17, 2006 4:45 AM
Think of it this way. VoIP phones allow you to place a phone anywhere
that a network connection exists.
Your Asterisk box will be on the network and will be easily accessible.
FXO and analog phones require point to point termination.
Phone to FXO. Period. What a pain!
VoIP phones are
Or FXS... Whatever. The point is port connect directly.
No one spam me on this one... 8)
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, January 11, 2006 2:55 PM
To: Asterisk Users Mailing List - Non
I think what he means is that an * server can support hundreds of phones
because the server connects to the network via a NIC.
Port count becomes irrelevant when you thing about VoIP phones
connecting to a VoIP server. They connect over the network not point to
point.
It is just a matter of
a PRI T1 line? Run FXS ports to
each fax machine and the TDM card will convert the digital T1 to analog
for faxing?
I have no POTS lines, just a T1 (PRI soon if I find out I can use
asterisk for regular POTS-type faxing).
Begin Original Message
From: Wiley Siler [EMAIL PROTECTED]
Sent
The consensus is usually a big NO though some have made more than 2
cards work.
I ran my system with 2 four port TDM cards and it worked fine. Others
have had nothing but problems.
This has to do with IRQs and the PCI bus if memory serves. A quick
search should yield info on IRQ and TDM cards.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Wiley Siler
*Sent:* Tuesday, January 03, 2006 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users
Just to make it easy, I will be reading the caller list from a another
server via a web page, parsing it and dialing.
After each pass, I just post back to the server web page and it updates
the other system.
Our tech just needs to review the log once daily.
W
-Original Message-
From:
: [Asterisk-Users] Dialer
On Fri, 2006-01-06 at 11:45 -0700, Wiley Siler wrote:
Just to make it easy, I will be reading the caller list from a another
server via a web page, parsing it and dialing.
After each pass, I just post back to the server web page and it
updates the other system.
Our
Title: Dialer
Hello All,
I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear.
First off let me premise this with no, this is absolutely not for doing call marketing.
I need to make my Asterisk box call a
What is your port density requirement?
For 24 ports the LinkSys SRW2024 is awesome.
They street for less than $500 and have good QoS.
For a smaller switch, they make a 12 port variant.
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent:
Applications
Wiley Siler wrote:
What is your port density requirement?
For 24 ports the LinkSys SRW2024 is awesome.
They street for less than $500 and have good QoS.
For a smaller switch, they make a 12 port variant.
Does the SRW2024 support port mirroring? I was shopping around, but
couldn't find
I recommend checking the following site...
www.voip-info.org
Lots of info for you there...
By VoIP phones, I think you are meaning soft phones which are software
based.
You will need a headset for the PC that runs the software phone.
Usually Logitech or Plantronics at about $50 a headset.
If
Title: 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
___
--Bandwidth
Title: WCFXO and T1 PRI Card?
Well, partial success so far Here is my
ztcfg
SPAN 1:
ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel
map:
Channel 01: FXS
Kewlstart (Default) (Slaves: 01)Channel 02: FXS Kewlstart (Default) (Slaves:
02)Channel 03: FXS Kewlstart (Default)
Title: WCFXO and T1 PRI Card?
I am getting an error about a broken pipe when I run
asterisk -vvvc
It reads zapata.conf as Found then dumps this error about a
broken sound pipe?
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
SilerSent: Thursday, October 06, 2005
No doc but I can tell you that the easiest thing to do is use a config
file and ftp if you have the ability.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Wednesday, October 05, 2005 2:13 PM
To: Asterisk
Subject: [Asterisk-Users]
Michael,
This is the list for Asterisk not Asterisk at Home.
That list list can be found at the same place you downloaded the AAH
software.
www.voip-info.org is the location of
Asterisk Wiki.
Now. On to your problems.
1. No
one can help you with that problem since we don know what kind
Assuming you can purchase online, just go to
voipsupply.com.
http://www.voipsupply.com/index.php?manufacturers_id=13
The switch between analog and digital makes a huge
difference to port density. With an analog
TDM card you can get 4 FXO/FXS ports per card.
With a digital T1PRI card, you
I got right in just fine...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters
Sent: Friday, September 16, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wiki down?
I'm unable to connect to
LOL - Congrats!
$30 down...
Let's see... how much to go?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Monday, September 12, 2005 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
Title: First PRI Installed - WOOT
Today I got my first PRI installed. It literally took less than 5 minutes and the circuit was up and we were making calls. The T100P is performing excellent. The Linux/Asterisk box is running well and the quality is great. The line is from MCI and they did a
Pay the license fee and get the GSM codec would probably be best.
The fee is nominal and the codec is a good one...
$0.02
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Hajek
Sent: Thursday, September 08, 2005 1:50 PM
To:
: RE: [Asterisk-Users] voice over atlantic
Probably missing something here. Never heard of GSM commercial licence
for asterisk.
Do you have any URLs?
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, September 08
: [Asterisk-Users] voice over atlantic
Yep. Thats G729, not GSM.
Btw, GSM codec implemented in Asterisk is EFR?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Friday, September 09, 2005 12:08 AM
To: Asterisk Users Mailing List - Non
Google can translate if that helps...
w
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
ShipmanSent: Thursday, September 08, 2005 4:44 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] sending fax
Thanks, but I can't read
Last time I talked to them, it was supposedly going to be released in
June... Then July,... Then August...
These are still vaporware as far as I can tell... If anyone knows of
anything different, I would love to hear it...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I have one and it is absolutely awesome. Works great and the quality of
Polycom conference phones is excellent regardless of protocol.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, August 26, 2005 9:50 AM
To: Asterisk
Title: PCI 2.3
Hello All,
Anyone know if this is backwards compatible with 2.2?
Here is the spec from the Mobo I am looking at.
Five 32-bit v2.3 Master PCI bus slots (support 3.3V/5V PCI bus interface).
Thanks!
Wley
___
--Bandwidth and
Bad URL... Too many R's in there... Correct...
http://www.voipzoneenterprise.com/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Watters
Sent: Thursday, August 25, 2005 10:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Just because you cannot get it to work does not mean that
IT does not work.
Just using the right motherboard is not enough. Did
you check for IRQ problems? You don't mention whether you have checked for
this.
Look for a thread called "Asterisk-Users Small office
setupusing analog lines w
What good does RAID give you on writes? None whatsoever. RAID only
helps performance on reading.
Come again? Writing to multiple hard drives in parallel is way faster
than writing the same file to one HDD.
You should Google the words RAID and Write Performance.
I assume you must have meant
Did you recompile everything * after your upgrade?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Stahl
Sent: Monday, August 22, 2005 10:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk 1.0.7 won't run after
Title: CRM software
Go look at the Asterisk @ Home install to see how they got
Sugar CRM integrated. It is a good start point and you can build from
there.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
ArcherSent: Thursday, August 18, 2005 8:29 AMTo: Asterisk
There are a dozen Linux based methods ranging from. Personally I like
the Mandrake offering called Multi-Network Firewall. It is pretty
turnkey and they have it available for download. It also supports
bonding which allows you to use multiple nics bonded together and views
as one connection.
Just Google the archive on 'IRQ issues'.
You can pretty much bet that 6 TDM cards on 6 PCI slots would suck
hugely.
Unless echo is your goal, you are not going to be pleased.
If you have to use 24 existing POTS lines, look into a channel bank and
interface it to a T1 card.
If you are planning
I use a DSP 500 and I love it. Great sound, good price.
IaxComm is hands down the best softphone I have found.
As you can guess it is for IAX though...
Cheers,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Monday, August 15,
I think the easiest way to tell if you don't get an answer is to see if
it uses IRQ sharing and if it allows you to assign IRQs individually.
A check of the BIOS instructions for that Mobo should be available at
the manufacturer.
W
-Original Message-
From: [EMAIL PROTECTED]
Also check out this getting started page
http://www.oneunified.net/support/asterisk/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent: Tuesday, August 16, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
And no RJ45 connectors? Doesn't sound like an IP phone at all.
Sure you did not get a phone for a Polycom PBX solution of some sort?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, August 16, 2005 9:38 AM
To:
Alejandro...
Go search the archive... There are tons of posts regarding Dell equipment
Here is how to do so if you do not know...
Go to www.google.com
Enter the following...
site:lists.digium.com Dell Poweredge
Thanks,
W
-Original Message-
From: [EMAIL PROTECTED]
]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Travers
Sent: Saturday, August 13, 2005 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Firewall will definatelyincrease
jittersinyourvoice conversation
Wiley Siler wrote:
The question was not can I
Do you mean this occurs when traffic is passed over an IPSec tunnel or
that it occurs anytime a tunnel is use on a machine that also is passing
VoIP traffic (outside the tunnel)?
I assume you must mean over the tunnel but I am curious...
Thanks,
Wiley
-Original Message-
From: [EMAIL
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Esben
Stien
Sent: Thursday, August 11, 2005 5:31 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Firewall will definately
increasejittersinyourvoice conversation
Wiley Siler [EMAIL PROTECTED] writes:
firewall (be it software
Of Wiley
Siler
Sent: Wednesday, August 10, 2005 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Firewall will definately increase
jitters inyourvoice conversation
Lokesh,
While adding a firewall may add a tiny bit of latency (non-noticeable
That should not be a problem. My users conference
using a voip line from an ITSP so at any time there may be 4-8 calls passing
over the firewall and terminating in the MeetMe conference. It works
great. I would recommend Pix BTW. Linksys would be my next
rec. But hey, they are both Cisco
Lokesh,
While adding a firewall may add a tiny bit of latency (non-noticeable by
the way) it in no way means you are gonna get jitter. An over utilized
data line might cause that but a firewall in and of itself will not. I
use a Pix to route my VoIP to an ITSP and I could not be happier. To
right. It would be dangerous not to have a firewall
for security reasons.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, August 10, 2005 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
Go to the wiki and search on SIP and NAT
www.voip-info.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jonny
hashem
Sent: Wednesday, August 10, 2005 1:24 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] does SIP works behind the
, I don't think there really are any major
differences. I've had no problems with ni1 and ni2 with Asterisk.
--
Tom Hayden
Astoria Telecom, LLC
www.astoriatelecom.net
On 8/9/05, Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
I am getting my first PRI installed in a couple of weeks
Title: 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
Switch to IAXCOMM and use an IAX extension. Problem
solved.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
KronstadSent: Friday, August 05, 2005 7:03 AMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Asterisk -
Caveat Emptor
Considering how he been as a list participant, I would be wary but it is
your dime...
Hope it works out...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Wednesday, August 03, 2005 1:44 PM
To: Asterisk
You may want to try a little research here...
www.voip-info.org
www.digium.com
Google: site:lists.digium.com asterisk process
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Karl
Sent: Tuesday, August 02, 2005 10:01 AM
To: Asterisk Users Mailing
It would not hurt for you to realize that
this is the Asterisk list and not the Asterisk @ Home forum.
AAH is a specifically configured turn key
product that someone was nice enough to package for people who dont want
to hand code their configs.
Thusly, it is not really something that
Go into the CLI on the box and type:
sip show users
sip show peers
Did you get two lists? One that shows the sip accounts and the other
that shows the registered sip accounts?
What does this show in the CLI: zap show channels
W
-Original Message-
From: [EMAIL PROTECTED]
Something very very different.
AAH is a package of Asterisk (1.0.7 I think) and AMP and FOP and other
tools...
Asterisk is the core software that runs in AAH.
So, Asterisk is the REAL software nuts and bolts while AAH is a nice
packaging of tools with Asterisk as the core.
You should go to
Did you build it using the 64 bit CentOS or another Distro?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Dobrin
Sent: Friday, July 22, 2005 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Do some debug on the calls and see what
you get.
W
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA
Sent: Thursday, July 21, 2005 2:56
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] T1 -
incomplete calls
Hi All
For the fella who wanted MOH music.
Royalty free stuff can be found here.. The Acoustic Guitar
is a nice collection
http://www.freeplaymusic.com/
Cheers,
W
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Right, so before resolution can be
attempted or had, you lash out on the forum.
If they had told you to stuff it or had
just ignored you, you might have something to complain about.
You are pissed that the ATA is not web
configurable? How in the hell is that VoipSupplys fault? You bought
Hmmm.. My nufone account is still running although it had problems
yesterday.
Seshu, try contacting Jeremy at nufone dot com. I think that is his
email at least. Last name should be Macnamera (sp?) I think. You can
search the archive for his name along with nufone.com if you need
other
freeplay.com i think. i will vverify for the url tomorrow at work.
the acoustic guitar stuff is nice...
Cheers,
W
From: [EMAIL PROTECTED] on behalf of Jim Archer
Sent: Tue 7/19/2005 5:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Use to providers for the call, pay two providers for the call.
You have two call legs so you are using two channels bridged at your *
box.
You will have to pay for those to legs...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of code
select
Sent:
This sounds like DISA which is great for saving bucks on LD if used
right...
You will still need two channels and thus it will still cost for both
legs...
Nature of the beast...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Latham
Sent:
I assume ISDN accomplishes this since the PRI is set to use channel 24
for signaling. Your 64K channels is data and the control overhead is
sent on the signaling channel.
Actually, everything I have seen is around 80K full duplex for a uLaw
channel with overhead. That is point to point...
W
Let me expand on the bandwidth point HTH made and maybe shed light on
your requirements
A 100baseT switched (no hubs) network has a lot of bandwidth when you
think in terms of VoIP. The uLaw stream (uncompressed) from an IP500
phone to the Asterisk box is not going to take more than 80K of
Depends on what you mean by expanding your network. Do you need a bunch
of new routers? Probably not. Do you need to consider port count at
every station? Absolutely. However, there is good news and bad news.
The good news is that most of the phones that are being recommended to
you actually
No idea on the phone ports but I doubt it as 100Mbit is sufficient and
the parts are cheap for the makers of the phones.
Not a bad switch but since you get 4 ports (one is used for connect to
wall) you may want to just up for the 8 port unless you know only two
people will use each switch. You
You clipped the original so there are some other things that need to be
known.
How many users are being supported again?
The biggest hits to servers seems to be due to transcoding in most
cases.
Look on the wiki for an explanation of server sizing and decide based
upon how you will connect your
I think one thing you may want to remember is that porting numbers to a
VoIP provider can make them EXTREMELY hard to ever port back to a normal
telco provider. Also, if there is ever a problem with the VoIP provider
(which has been common lately) then you are in deep trouble. For a
mission
Sounds like a PRI T1 will be fine for you to start with. It offers you
23 voice channels (one channel is used for signaling).
That means you can get a single Digium T1 card for around $600 or you
can get a quad T1 card for around $220 (with echo cancellation). If
there is no move to expand,
Great points but I think the ease of config on Polycom via FTP along
with the ease up firmware updates is a real winning combination. I have
yet to need the kind diagnostics you refer to while troubleshooting. I
copy a valid config, change the values as needed and load it to the FTP
server. Boot
Hello and welcome...
Most of what you want to know is available on the wiki located here...
http://voip-info.org/tiki-index.php
Just scroll down to the All Things Voip section.
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Pastore
Read directly off of one of my phones power supplies...
AC Adaptor
I.T.E Power Supply
Model: AD41-1200400DU
Input: 120VAC 60Hz 200mA
Output: 12VDC 400mA
P/N: GJE-AD41-995 LEVEL 3
Outer ring is negative, inner is positive
There should be more in the manual for that phone I assume
]]
On Behalf Of Wiley Siler
Sent: Tuesday, July 05, 2005 4:53
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
memtest at the cd boot
prompt)
--Rob
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Wednesday, 6 July 2005 10:45
AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Epia
C3 Linux
OK. Something is truly
This did wind up being a matter of memory...
Thanks,
Wiley
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, July 06, 2005 10:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Epia C3 Linux
On Tue,
...no external drivers required
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 2:42
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Epia
C3 Linux
Did it require any special work or did you
just download the ISO for FC3 and install
OH, yes, the error is always Segementation
Fault when I try to write the ext3.
W
From: Wiley Siler
Sent: Tuesday, July 05, 2005 4:53
PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Epia
C3 Linux
I have attempted FC3
Of Wiley Siler
Sent: Tuesday, July 05, 2005 4:53
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
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
:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Epia C3 Linux
Wiley Siler wrote:
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
Discussion
Subject: RE: [Asterisk-Users] Epia
C3 Linux
I have Fedora Core 3 running great on an
Epia mobo
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 12:54
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Epia C3 Linux
C3 Linux
It installed directly from the FC3 dvd, no
changes...no external drivers required
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 2:42
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Epia
C3 Linux
Did
Wow, found the papers out at Broadband. Even more shocking than
expected!!
Papers located at bottom of page here if anyone wants them:
http://www.broadbandreports.com/forum/remark,13748234~mode=flat~days=999
9~start=20
I long wondered what the link between Brandon and Pamela was.
These guys
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