http://nlug.org/mail/nlug__2003_12/0094.html
Kevin
-Original Message-
From: Panny Malialis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 2:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
I cant wait to see the asterisk on an xbox
Regarding using asterisk as a T-Bird replacement, that would be pretty
cool, as asterisk could do some incredible monitoring and call
generation tricks.
But, bugs in the PRI driver software (especially beefed up error handling)
would have to be fixed first, as asterisk doesn't handle high volume
While the rest of you were chatting about the smallest * server, I was
sitting her staring at the telephone hanging on the wall.
It is a Western Electric set in a varnished pine box with an earpiece
you hold in your left hand and a mouthpiece attached to the box. You
crank the magneto with
What is the best inexpensive voip phone out there? I want to try
a few with *, but don't want to go broke while I'm just playing
around...
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box 726
http://nlug.org/mail/nlug__2003_12/0094.html
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Panny Malialis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 10:58 AM
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
I cant
Hello
when i dial a toll free no using sipphone
i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on
'Zap/2-1' -- Executing SetCallerID("Zap/2-1",
"17473863282") in new stack -- Executing
SetCIDName("Zap/2-1", "Deepak
While the rest of you were chatting about the smallest * server, I was
sitting her staring at the telephone hanging on the wall.
It is a Western Electric set in a varnished pine box with an earpiece
you hold in your left hand and a mouthpiece attached to the box. You
crank the magneto
Tim Sailer wrote:
What is the best inexpensive voip phone out there? I want to try
a few with *, but don't want to go broke while I'm just playing
around...
Tim
AFAIK Grandstreams are still the cheapest..
___
Asterisk-Users mailing list
[EMAIL
Well, the nice things about telephones, in the US anyway, is that they are
generally backward-compatible with each other.
Why not, as a first step, connect a normal telco line to L1 and L2, and see
if you get dial tone through the receiver?
Scott M. Stingel
Emerging Voice Technology Inc.
Palo
If you've got a Linux workstation, www.vovida.org offers
sipset, a free softphone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Tim Sailer
Sent: Tuesday, February 03, 2004 4:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voip phones
ON second thought:
I re-read your message - missed the part about the magneto. Maybe you
shouldn't connect that phone to the PSTN after all! (or to a digium card)
Magneto-generated ring detection is a bit beyond the digium card spec I'm
sure!
Cheers!
Scott M. Stingel
Emerging Voice
I just followed those steps and its working wonderfully. I did have to
download and compile asterisk-addons from CVS, as well as the steps
illustrated above. Now I'm going to make some PHP Web Reports so the
users can view the CDR.
-Original Message-
From: [EMAIL PROTECTED]
I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue
telephone adapter
Why?
- brilliant user interface, with or with out a web browser
- cristal clear voice even with low band codecs
- PPP over ethernet (PPPoE) aware
- continual firmware improvement
- plenty of tweak options
On Tue, 3 Feb 2004 11:33:24 -0600, David Gomillion wrote
Steven Critchfield wrote:
On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
[flames, non-flames etc. snipped]
what about something like this? NOTE:
and with the HT-286 you get a Chinaman in a box! :)
- Original Message -
From: Michael Koehler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 5:45 PM
Subject: Re: [Asterisk-Users] voip phones
I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286
I am a new asterisk user and I love what I see so far.
I have a question about distinctive ring though.
In my situation, we have 1 phone number for voice calls and one for faxes.
They share the same line, and right now I use vgetty with mgetty+sendfax and
VOCP to deal with calls and faxes.
Vgetty
you can do this with MeetMe, but you don't have to. you can also use
Parking, which makes things a little simpler.
in either case, the strategy is going to be something like:
1. Record the soundfile
2. Park the inbound caller
3. Use a .call file or the manager interface to initiate an outbound
I've got a dumb Western Electric payphone and some homebuilt hardware to control the
coin relay which is accessible to Asterisk through the AGI interface. I'd like to be
able to set the state of the coin relay to collect at the end of a call if a called
party answers.
[Hey, I admit this
Just for fun, try this:
exten = 922,1,Flash(Zap/1)
exten = 922,2,Dial(Zap/1/*022)
exten = 922,3,Congestion
exten = 922,4,Hangup
and see if it gives the same error. I'd be interested to see if
there's perhaps some strange variable swapping going on.
I gave that a try, but the same
On Tue, 3 Feb 2004, Scott Stingel wrote:
Well, the nice things about telephones, in the US anyway, is that they are
generally backward-compatible with each other.
Why not, as a first step, connect a normal telco line to L1 and L2, and see
if you get dial tone through the receiver?
I'm sure
Yes, I hereby withdraw my suggestion (see my earlier, 2nd post)!!
NOT advocating connecting a magneto to the telco circuit! and only to a
digium card if you want to let off some steam (and probably fry the
card)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
not to mention, fortune cookies are included! :)
Hey Chinaman...
I was wondering if the following SIP phone is just a Grandstream's OEM
or just a japanese copy...
http://sipphone.livedoor.com/
What do you think?
Isamar
___
Asterisk-Users
Does the voicetronix card work with Asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, February 02, 2004 11:06 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from
digum
Well, the nice things about telephones, in the US anyway, is that they are
generally backward-compatible with each other.
Why not, as a first step, connect a normal telco line to L1 and L2, and see
if you get dial tone through the receiver?
I'm sure that your LEC wouldn't minx 100
Will this product be available in the next few weeks?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Saturday, January 31, 2004 3:01 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 8 lines - best approach
How about a 16 port
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gene Kochanowsky
Sent: Wednesday, 4 February 2004 12:18
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pictures of new multiport
FXO/FXS from digum
Does the voicetronix card work with
Thud!
Will there be no FXO daughter boards for
the TDM400?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Gene Kochanowsky
Sent: Thursday, January 29, 2004
7:55 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] TDM400
FXO???
Any word/news on the
Hi Folks,
I recently setup an asterisk system in order to
provide a telephone phone system for my web hosting business at a very low
expense. My problem is that DTMF tones are not being recognized when calling the
IPKall phone number. Calling my server via FWD and IAXTel works out fine
Does the voicetronix card work with Asterisk?
Yes, and no.
It works in that you are able to use it to make and receive calls - but to
say it works well would be an overstatement.
We are currently using the OpenLine4 card and are having problems dialing
(card dials too early ; doesn't support
I don'thave a DSL filter, Butall my telephones do have filter and work very well. I'll try to use a filter. Thank you all.
Best,
Michael
=
Perhaps you forgot to put a filter between the line and your X100P?
-Tilghman
Eric Wieling [EMAIL PROTECTED] wrote:
Do you have a DSL filter
Will there be no FXO daughter boards for the TDM400?
There will be. Units are again in production after having an issue that
had us stuck for about 2 months.
Mark
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Asterisk-Users mailing list
[EMAIL PROTECTED]
I am in Japan and I was just going around in some
shops in the web...
Isamar
I found a site somewhere that referenced the livedoor sipphone to:
LivedoorSIP phone terminal development original page
Http: //www.grandstream.com/y-product.htm
The manual it is
Which means Livedoor sip phone
Joshua,
I've been looking into doing the same for my biz as well. I haven't
heard of IPKall and perhaps they aren't setup for what you want to do.
If this a vital part of your business I'd consider using a commercial
IAX provider to give # a toll free or local # for users to call in. If
you want
The photo of the phone says Grandstream Budgtoe 100 when you
click to see the larger image of the phone the text on the
buttons becomes clear.
They look to be selling aservice and the phone to go with it
but I'd have to as my wife. She's fluent in Japaneese. I'm
not even close.
Hey
I have my asterisk box on the public network. I have a winders box on the
public network, running diax. I have a winders box, same setup, behind
my Linux iptables firewall, on a private network. Both boxes cann register
iax2 to asterisk, and dial, but as soon as asterisk tries to do the native
Greetings,
It appears you are correct, as a test I just set it
up so when an incoming call came in it dialed Tellme and their system didn't
pick up on the DTMF tones either. I guess I will have to wait for my other phone
number to be setup.
- Joshua Colp.
Joshua,I've been looking into
Wehave (2) cartons of (56) AC Power Cubes for
the Cisco 7905, 7910, 7940 and 7960 IP Phones.
These are brand new, and include the power
cord.
They come with a 1 year warranty.
Cost is $17/ea, minimum order of 10
pcs.
Cory
Andrews***b2 Technologies454
I've looked online through both google and bugs.digium.com and cannot seem
to find this problem anywhere, so i'll ask its unpatched source code for
both and everything else works fine.
Does anyone else have the issue of asterisk dying with no messages when
trying to transfer a voicemail from one
At 6:26 PM -0500 2/3/04, tad wrote:
you can do this with MeetMe, but you don't have to. you can also use
Parking, which makes things a little simpler.
in either case, the strategy is going to be something like:
1. Record the soundfile
2. Park the inbound caller
3. Use a .call file or the manager
At 8:31 PM -0700 2/3/04, Rohde wrote:
I've looked online through both google and bugs.digium.com and cannot seem
to find this problem anywhere, so i'll ask its unpatched source code for
both and everything else works fine.
Does anyone else have the issue of asterisk dying with no messages when
Thanks for the reply Mark. When do you expect to ship and are you taking
orders?
Gene
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Tuesday, February 03, 2004 9:40 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] TDM400 FXO???
I've got a dumb Western Electric payphone and some homebuilt
hardware to control the coin relay which is accessible to Asterisk
through the AGI interface. I'd like to be able to set the state of
the coin relay to collect at the end of a call if a called party
answers.
[Hey, I admit this
How do you start asterisk? using safe_asterisk? or what cli options do
you give it?
bkw
On Tue, 3 Feb 2004, John Todd wrote:
At 8:31 PM -0700 2/3/04, Rohde wrote:
I've looked online through both google and bugs.digium.com and cannot seem
to find this problem anywhere, so i'll ask its
Before I got into my question for the day I'd like to applaud all the
helpful folks and time spent behind the asterisk project to get it where
it is today. Great work and between this list, the doc list and the irc
channel it's been a pleasure to deal with people willing to help others
when and if
We recently took a few Polycom Soundpoint IP 500
to test out in Asterisk environment. So far it has been good. Call
Hold, Transfer, DMTF etc.
However, I do notice every now and then the
Polycom fails to register with Asterisk. Asterisk console outputs the
following:
Feb 3 13:02:32
So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to
the point where it needs to be unplugged, due to software errors.
This is a first.
My suspicions are that this bug in Asterisk is causing the lockups:
http://bugs.digium.com/bug_view_page.php?bug_id=889
It seems unusual
The majority of sip to pstn gateway providers (vonage, voicepulse, and
others) appear to be setup for a one line only type of set up. Their web
sites seem to be heavily geared for these one line setups.
Anyone willing to comment on what type of pricing plans these providers
offer when using iax2
an associate of mine sent me an email of the slick sheet on this one. I
understand that mentioning this vendor has resulted in some flamethrowing
on the list, and I do not want to cause trouble - just looking for some info.
Thanks!
Sam Z
___
I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i
pushed 5 calls i'd be charge per min for each call. Granted both the
companies above cater to * quite heavily.
On Wed, 2004-02-04 at 01:40, Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage,
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