> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Andrew Nelson
> Sent: Thursday, November 27, 2003 9:39 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Modem cards??
>
> Ok if standard voice modmes do not work with Asterisk as an FXO then why
ning go to mlpppd for data.
I'm guessing here but I assume you would have a hunt group assigned to the
voice pri channels for inbound voice calls. If you wanted inbound data you
would need a second hunt group assigned to the data channels.
Is that a good guess Walker? :-)
John Breeden
Plum Hall, Inc.
Hawaii
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or the budgetone is:
[jrb]
type=friend
host=dynamic
username=jrb
secret=x
dtmfmode=rfc2833
context=home
reinvite=no
canreinvite=no
qualify=1000
I can't find a solution to in the archives and I've looked at all the
documentation I can find setting up the budgetone on *.
Any po
he budgetone is:
>
> [jrb]
> type=friend
> host=dynamic
> username=jrb
> secret=x
> dtmfmode=rfc2833
> context=home
> reinvite=no
> canreinvite=no
> qualify=1000
>
> I can't find a solution to in the archives and I've looked at all the
> documentat
I've come to the same conclusion. There seems to be very little specific
info in the archives about what vendor's products work and what doesn't.
Could someone post what FXO sip gateways work great with * ?. It'll save
return shipping charges :-)
Thanx
John Breed
http://news.google.com/news?hl=en&edition=us&q=at%26t+voip&btnG=Search+News
http://www.att.com/news/item/0,1847,12627,00.html>
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These are just claims.
Post a pdf of the contract.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of John Brown
> (CV)
> Sent: Friday, December 12, 2003 8:21 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SIPURA Breaches Contract
>
> 2. SIPURA an
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Hector
> Q.-datafull
> Sent: Friday, December 12, 2003 4:33 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] WARNING[1125350192]
>
>
> Can anybody help me with this error?
> I use * only for voip w
Kill all unneeded processes, this is all I'm running on my * boxen.
[EMAIL PROTECTED] asterisk]# ps ax
PID TTY STAT TIME COMMAND
1 ?S 0:06 init [3]
2 ?SW 0:00 [keventd]
3 ?SWN0:00 [ksoftirqd_CPU0]
4 ?SW 0:00 [kswapd]
5
Am I
assuming that a GS set to early dial to * dosn't work. Or am I missing
something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's
not/supported or GS bug, only asking because it's assumptions that alwas get me
:-)
GS
firmware 1.0.4.26
Than
How is Vonage doing it?
http://www.vonage.com/features_fax.php
John Breeden
Hawaii
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> Underwood
> Sent: Sunday, December 14, 2003 2:24 PM
> To: [EMAIL PROTECTED]
> Subject: Re:
competing against them too. If I was vonage I'd be telling
the world how important a ip fax line was :-)
Second, the residential/soho market almost demands
replacement of analog with voip. It's almost impossible to justify the roi
unless you do.
Marketing Hat Off
John
Breeden
H
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of James Sharp
> Sent: Sunday, December 14, 2003 6:01 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
>
>
> > It's just my lowly opinion but I too must a
I'll just throw this out. I've had some MB's in the past with flaky support
for APIC. It resulted in weird interrupt problems. Disabling APIC and a
kernel recompile solved it.
John Breeden
Hawaii
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PRO
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of WipeOut
> Sent: Monday, December 29, 2003 9:57 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] fedora core 1 install problem
>
>
>
> While your work around is fine it does add one more thing
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Nicolas
> Bougues
> Sent: Tuesday, December 30, 2003 9:18 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Backup Proxy & Automatic Failover
>
>
> On Tue, Dec 30, 2003 at 06:49:51PM +, Adthr
Is there anyway to append the '#' symbol to a dial string? - hex/octal
whatever? I'm surprised that I can't find anything searching the wiki or
google.
Thanx
John Breedn
Hawaii
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Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how you
can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? - hex/octal
whatever? I'm surprised that I can't find anything searching
Thanks!
Right syntax - wrong box :-) (inter-iax between to *s - needed to apply
the suffix to the box talking directly TO the zap channel ... duhhh .)
Caught yet again by my own wrong assumtion
Eric Wieling wrote:
John Breeden wrote:
Been there, done that - no joy :-)
It appears the
http://www.grandstream.com/user_manuals/budgetone100.pdf
Mike Chapman wrote:
Hi-
I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to
make sure it has dhcp enabled and it is. The display says no
Very cool ...
Have you tried to compile it against Mono?
-JB Hawaii
Thorben Jensen wrote:
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Henry Devito
Sendt: 16. marts 2005 16:17
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] IPS
<- don't have
Mar 18 19:29:55 in.tftpd[32679]: RRQ from 192.168.1.13 filename
P003-07-1-00.sbn <- have
Righ now the cisco akes a damn fine sexy door stop.
Thanx in advance
---
John Breeden
Plum Hall, Inc.
Kamuela Hawaii
_
ge-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
John Breeden
Sent: Saturday, March 19, 2005 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Yet another cisco 9760 7.x
firmware failure
Got my first Cisco 7960G
---
John Breeden
Kamuela Hawaii
John Breeden wrote:
Got my first Cisco 7960G from voipsupply this week. They sent me the
sip firmware upgrades through 7.3.
All went well through 6.X, Then the 7.X firmware upgrade flash.
(7.1.0). Yea, I read the archives and was forwarned :-)
The Universal A
ally nessessary, but I had 'um so
I did 'um.
Word of caution: the 7.x firmware sent to me by voipsuppy was incomplete
as to the 7.x releases. They left out the .loads files. Without those
files you'll end up with the dreaded "application load failure".
BTW: The servic
It dosn't run under the mono framework. There, now you have an answer :-)
Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
I am not sure that it will run on Mono, for now I only support it on
Windows. (I will test it on Mono later).
One thing (or so) at a time is indeed a
Asterisk runs just fine under Workstation, GSX and ESX but only with
ztdummy.
Paul Fielding wrote:
Anyone tried running Asterisk in production on a vmware box? I'm considering
the possibility and looking for sucesses/failures...
regards,
Paul
-
I'm running a pair of these. Both run Vmware ESX and one virtual machine
runs * using only ztdummy. Seems to run just fine. It's not not used in
production, just test/development.
Don't know how much that helps you.
Geoff Nordli wrote:
I am looking at using a dual Xeon Dell 1850 with a PERC 4e/S
For the *Brave At Heart* it might run under wine/xoveroffice.
ms net framework 1.1 appears to run in xoveroffice, don't know about
2.0beta: http://www.interex.org/hpworldnews/hpw310/01lab.jsp
[EMAIL PROTECTED] is believed to have said:
It dosn't run under the mono framework. There, now you h
Stoopid question 1:
I see how to make a call but for the life of me I can't see how to DROP
a call.
Thorben Jensen wrote:
Release 0.66 of IPSwitchBoard is now available for FREE download at:
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA
Enhancements:
Support for Call Parking a
X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801.
John Simon wrote:
Is anyone using a net4801 and an analog only setup? I am looking for a
modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it
won't work with the net4801.
ke your "modem donation" to Digium? :-)
John Breeden
Hawaii
Matt Ryanczak wrote:
I ended up using a sipura spa-3000 for FXO/FXS. It works great.
http://www.sipura.com/products/spa3000.htm
-Matt
On Mon, 2005-03-21 at 09:40 -0600, John Simon wrote:
This is the same experiance I had with
.. :-)
Ah yes, I'ts currently only a dream:
Soekris with 2 T1s, 4 fxo/fxs ports and gsm running to 20 cisco
eye-candy phones All from this itty-bitty boxen -)
John Breeden
Hawaii
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
Strange;
It works for me. The x100p (Digium
If you answer the question why 95% of the desktop market is owned by
Microsoft or why gasoline is used as the fuel for internal combustion
engines, you will know the answer as to "why sip".
The "best" technology dosn't always win the market.
-JB
Andrew Kohlsmith wrote:
On March 21, 2005 01:07 pm
My x100p is keyed only for 3.3v ... interesting.
Matt Ryanczak wrote:
It works for me. The x100p (Digium 100 buck model) I have is slotted for
3.3v and works fine.
My X100P cards are both clones and they do not work in my Soekris boxes.
I even fiddled with power, pushing the Soekris to th
For what it's worth, here's the chipset (tiger320):
http://www.tjnet.com/software/download/data_sheets/Tiger320_data_sheet.pdf
Matt Ryanczak wrote:
It works for me. The x100p (Digium 100 buck model) I have is slotted for
3.3v and works fine.
My X100P cards are both clones and they do not work
Everything's for sale . if the price is right . STARTING BID?
:-)
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
It works for me. The x100p (Digium 100 buck model) I have is slotted
for
3.3v and works fine.
My X100P cards are both clones and they do not work in my So
for the 4801 ... Oh well.
---
John Breeden
Hawaii
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
Everything's for sale . if the price is right . STARTING BID?
:-)
u start... :)
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FYI - new soekris boxes ..
Original Message
Subject:Re: [Soekris] net5801 & net7501
Date: Wed, 23 Mar 2005 15:34:20 -0800
From: Soren Kristensen <[EMAIL PROTECTED]>
Organization: Soekris Engineering
To: Jabbar Fagan <[EMAIL PROTECTED]>
CC: [EMAIL PROTECTE
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 03/24/2005 06:42:24 PM:
Now, how do you know if the wiring is truly Cat3? Just because the raw
wire is Cat3 means nothing if they wrapped it around a few fluroescent
lights... ;)
Acually, the first demo of 10baseT at Interop after the s
Rescue CD from Acronis for Linux
http://www.acronis.com
Works every time ...
Jeff Glassman wrote:
After getting my feet wet with [EMAIL PROTECTED], I want to set up a second
asterisk box to add a call shop billing and other add-ons such as LCR.
My question is as follows. Is there a backup program
CAUTION: voicemail screwed up for me (garbled) with upgrade to 23, went
back to .22 and all is well.
Don't know why, I'll look at it later.
dean collins wrote:
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/
Or directly from Grandstream at
http://www.grandstream.com/BETATE
Short of finding somewhere to tap 12v off the board that 1) would'nt
make the danged thing beep and 2) voiding the warrantee cdrom??) , I'd
just juryrig an external 12v supply along the lines of
http://www.soekris.com/PowerAccessories.htm.
I'm assumong the tdm400p only taps the 12V for RI and n
I love soekris boxes, but in my humble opinion the answer would be be no.
Just for yucks set up a 2-3GZ bix and compare it with a 4801, perhaps
you will com to a different conclution that I.
If some kind sole could port the codecs to use Serin's minipci
encryption card, than it might be a differ
This bit me too. Had to turn nat off on the 7960Gs
Kristian Kielhofner wrote:
Peter J VERNON wrote:
Guys..
I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and
have a couple of Cisco 7960s and a Grandstream phone.
I can make calls from the 7960. When I get a call placed to th
You don't need the xml files, they arn't supported until V 7.X and even
then aren't needed.
I avoid the whole bsd tftpd/inetd thingy altogether. It's always proved
somewhat flakey for me over the years. Instead I use atftp
(ftp://ftp.mamalinux.com/pub/atftp/) which can run as a standalone deamo
Might be a dumb question :-)
You DO realize it's SIP.cnf
and not the literal name "SIPmac_address.cnf" ?
Davin O'Neill wrote:
Thanks! That did the trick. My tftp server sent the OS79XX.TXT file and
the P0S30201.bin image file. The systems logs show:
atftp: severing OS79XX.TXT to Phone IP addre
y
conpare to GSX and ESX servers. Perhaps in the future
BTW: MS uses vmware inhouse :-)
John Breeden
Hawaii
public wrote:
Does MS Virtual server support running linux? Thought you could only run
windows on it?
Vmware's server products are apparently very good. Their Workstation
p
Benjamin on Asterisk Mailing Lists wrote:
In this respect, something like CoLinux and Astwind is definitely a
good thing because it lowers that entry hurdle, even if only the
perception thereof.
rgds
benjk
When NT40 first came out, the company I worked for ported all of it's
management products
Blackberry announced a sip based voip phone available early '05
http://www.blackberry.com/news/press/2004/pr-18_10_2004-02.shtml
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To UNSUBSCRIBE or up
Cat3 - which used to be called "D Inside Wire" (DIW) *is* the wire
spec'd in the 10baseT IEEE standard. The existing wire plant is
currently to the 10baseT standard., at least as far as the wire goes.
(It was originally invisioned that 10bt and analog/digital voice would
be running in the same
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