On Wed, 2007-08-22 at 08:50 -0500,
[EMAIL PROTECTED] wrote:
Date: Tue, 21 Aug 2007 21:01:50 -0400
From: David Cook [EMAIL PROTECTED]
Subject: Re: [asterisk-users] 99 bottles of beer
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Does anyone know if Asterisk Home (2.7) runs OK on a 64 bit architecture?Thanks.
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2ยข/min or less.___
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sk-users@lists.digium.com><asterisk-users@lists.digium.com>From: Al Lougher [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 11:29:41 -0700 (PDT)Subject: [Asterisk-Users] Asterisk home on VMWare time sync issues Hello - I am using AH on VMWARE. I have noticed th
Hello - I am using AH on VMWARE. I have noticed that the date and time periodically loses sync with the server system time. Does anyone know why this would be, or where if I need to change a setting somewhere so it keeps time properly?Thanks! Alan.
Want to be your own boss? Learn how
I've just completed downloading and installing [EMAIL PROTECTED] on my already
running
server, and came to view the config areas, however I already have Apache
installed, and operating on the server, so viewing port 80 takes me straight
to the relevant apache served page. Is there a way to
I just started another html process running as user asterisk and group
asterisk and changed the listen directive to another port. I am not
running asterisk at home, but freepbx which is very similar, so I am
told.
on Monday 04/24/2006 James Nunnerley([EMAIL PROTECTED]) wrote
I've just
I have installed asterisk @
home 2.6. I am using a Telasip VOIP account. When I make outbound or
inbound calls the calls seem to connect and then get hung up. I was
wondering if there was something that I am misisng. I have tried several
different sip.conf configurations. Here is what they
My guess, is nat problems. Just for fun, try dialing your inbound
number from something not connected to that asterisk box, say a
cellphone. I know you're using IAX and SIP, so you'd think you
wouldn't run into a double-nat problem (nat going out, nat coming in),
but you never know. I have odd
/9/2006 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
My guess, is nat problems. Just for fun, try dialing your inbound
number from something not connected to that asterisk box, say a
cellphone. I know you're
I have installed asterisk @ home 2.6. I am using a
Telasip VOIP account. When I make outbound or inbound calls the calls seem
to connect and then get hung up. I was wondering if there was something
that I am misisng. I have tried several different sip.conf
configurations. Here is what they are
Can anyone tell me how to recover an Asterisk
password if it is forgotten or do I have to do a complete
re-install.
Kris
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kchase wrote:
Can anyone tell me how to recover an Asterisk password if it is
forgotten or do I have to do a complete re-install.
Which password?
Do you have the SSH password for the [EMAIL PROTECTED] server?
David
Kris
the password to login into asterisk
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 3:11 AM
Subject: Re: [Asterisk-Users] Asterisk @ Home password
: [Asterisk-Users] Asterisk @ Home password recovery
kchase wrote:
Can anyone tell me how to recover an Asterisk password if it is
forgotten or do I have to do a complete re-install.
Which password?
Do you have the SSH password for the [EMAIL PROTECTED] server?
David
Kris
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 3:11 AM
Subject: Re: [Asterisk-Users] Asterisk @ Home password recovery
kchase wrote:
Can anyone tell me how to recover an Asterisk password if it is
forgotten or do I have to do a complete re-install
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the CID
only shows Analog Line on the Grandstream 2000XP phone. Does anyone have any
ideas even where to look to change this?? Is it a setting
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the
CID
only shows Analog Line on the Grandstream 2000XP phone. Does anyone have
any
ideas even where to look to change this?? Is it a
OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to
POTS lines. Everything seems to be working just fine, but I have some
questions on how to access voicemail options. I can leave a message for an
extension, but when I try to retrieve it by using *97 it asks for
What happens if you press *98 and enter the extension and password? are you
using speakerphone? tried it with the handset only?
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Monday, July 11, 2005 7:51 AM
Subject: [Asterisk-Users] Asterisk
It might be a DTMF issue. You may need to change the DTMF setting for
your extension. Can you make it work using x-lite?
Tom
On Jul 11, 2005, at 10:51 AM, [EMAIL PROTECTED] wrote:
OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B
interface to
POTS lines. Everything seems
Hank,
There is, look again on the [EMAIL PROTECTED] sourceforge site.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of hank
Sent: Wednesday, 29 June 2005 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
he must have just added it all I saw last I looked was the forums
thanks for this ifnormation
take care
hank
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 30,
he must have just added it all I saw last I looked was the forums
thanks for this ifnormation
take care
hank
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 30,
Hello I saw Ver1.2 is out. Whats new?
Thanks for the hard work, David
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Hello I saw Ver1.2 is out. Whats new?
Thanks for the hard work, David
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Hello I saw Ver1.2 is out. Whats new?
Thanks for the hard work, David
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New features include:
CentOS 3.5
Asterisk 1.0.8
New Zaptel Driver from CVS
Built-in DHCP server
David Shaw wrote:
Hello I saw Ver1.2 is out. Whats new?
Thanks for the hard work, David
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This is NOT the AAH list, please check sf.net for information
On Wed, 29 Jun 2005, David Shaw wrote:
Hello I saw Ver1.2 is out. Whats new?
Thanks for the hard work, David
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Hmmm..maybe the change log will reveal something...
Notes:
[EMAIL PROTECTED] - Taking calls in an hour
---
version 1.2 - 06/29/05
Features:
-
* Asterisk 1.0.8
* Flash Operator Panel 0.21
* Festival Speech Engine version 1.95
* weather agi scripts
what is the dhcp server used for?
- Original Message -
From: JD Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 29, 2005 5:49 PM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new?
there is no list for [EMAIL PROTECTED]
and why aint it aloud on this list its asterisk related aint it?
chill out guys
- Original Message -
From: Mike [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 29,
hank wrote:
there is no list for [EMAIL PROTECTED]
and why aint it aloud on this list its asterisk related aint it?
chill out guys
AAH is an abstraction layer for Asterisk, and the issues that relate to
it and not Asterisk belong on it's own list. If there is no list, which
would surprise
Hank, yes there is. Go to the sourceforge site, there is a forum just
for aah there.
Any specific aah questions should be posted there.
Any specific amp questions should be posted on the AMP sourceforge page.
Any general asterisk questions should be posted here.
Cheers,
Dean
-Original
Believe it or not, the AMP users list seems to accept aah issues. If
you take it there, Hank, you stand a pretty good chance of getting it
answered.
Darren Wiebe
[EMAIL PROTECTED]
Chris Mason wrote:
hank wrote:
there is no list for [EMAIL PROTECTED]
and why aint it aloud on this list its
perhaps I should create a list for [EMAIL PROTECTED] any one want me to do this?
I can create one on yahoogroups
I find email list easier to use then the forums and I am sure I am not alone
in this. let me know off list
[EMAIL PROTECTED]
if you want a [EMAIL PROTECTED] list started
laters
hank
there is no list on sourceforge I checked whitch is why I offered to start
one up sense it isn't aloud on this list.
once again folks let me know
thanks
hank
- Original Message -
From: Chris Mason [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
okay my apologies
thanks
hank
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 29, 2005 8:21 PM
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new?
As a newbie to Asterisk, I'm in love.
There is no information discussing the way to use the FTP program vsftpd
which is need for phone configurations. So far I've been able to add a user
with useradd and add that user to the VSFTPD.USER_LIST and now I can FTP to
my [EMAIL PROTECTED] server but
Paul,
1.) These questions are specific to [EMAIL PROTECTED], not asterisk, and
you probably ought to post your question to the forum at
asteriskathome.sourceforge.net (especially since [EMAIL PROTECTED] has specific
Cisco config options.)
2.) These questions have been answered before, look
I have [EMAIL PROTECTED] connected to broadvoice. Intermittently, when a call is placed to my asterisk box, the caller sometimes does not hear the ringing tone, but just silence. Although the call is going through. On the asterisk end, the phones ring, but the ringing tones in the callers phone
I ditched the idea of using Asterisk straight through my router, there
was too much to set up in too little time for me. I've found a spare
machine and installed [EMAIL PROTECTED] on it. Things run smoothly except
for connecting to my IAX provider; It doesn't even look like packets
are going out
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton
Humphrey
Sent: Friday, June 10, 2005 10:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] [EMAIL PROTECTED] connecting through firewall/router
I ditched
Yeh Tracy, I have a suggestions.why? Spreading out the information
(especially on privately managed and owned boards) is not the solution.
There is nothing on your site of value that couldn't be done on the
sourceforge site.
Dean
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I have created a new forum for [EMAIL PROTECTED] users (I guess it could
be used by any asterisk users). I am looking for suggestions on forum
categories. If you have any, I would greatly appreciate your input. I
have some already setup but I am not so sure of how logical the
categories are.
[EMAIL PROTECTED] 1.1 Released and can be downloaded from Sourceforge.
http://sourceforge.net/project/showfiles.php?group_id=123387package_id=135368
Cheers,
Max W. Blackmer
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With the addition of web based CRM into this machine, is there a plan to
implement traffic shaping in some way?
I know my users will be busy with the CRM side of this but I would not
want it to affect the RTP side fo things if too many users are connected
to the web.
Thanks,
Wiley
I don't see the SugarCRM being part of the install.
How do you activate this?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 31, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On mine it is http://ipofmachine then click the crm link
but I haven't found the default login and password for it yet.
Kanuri, Seshu (Company IT) wrote:
I don't see the SugarCRM being part of the install.
How do you activate this?
Seshu
-Original Message-
From: [EMAIL
I would also like to know about compatability of backups and restores.
Does this version restore from any of the other version? Basically 0.9
and higher?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent:
Just a guess if they followed old login/password
It could be...
Maint/password
Or not but worth a try...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Litwiller
Sent: Wednesday, June 01, 2005 2:16 PM
To: Asterisk Users Mailing List -
admin/password is the login info
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Wednesday, June 01, 2005 6:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 1.1b1 has been
We have replaced the simple contact management system
in [EMAIL PROTECTED] with SugarCRM a full CRM system. This
might seem like over kill for a home PBX but Sugar has
the best contact management we have seen. With click
to dial functionality and the ability to import data
from other contact
Can any one tell me what the mysql password, no its
not password.
[EMAIL PROTECTED] root]# mysql --user=root -p
Enter password:
Thx
Q
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Can any one tell me what the mysql password, no it's
not password..
Try passw0rd with a disgit ZERO not o
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Can any one tell me what the mysql password, no it's
not password..
Try passw0rd with a disgit ZERO not o
Try amp109
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thx
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: 30 May 2005 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED]
Can any one tell me what the mysql password, no
I recently installed [EMAIL PROTECTED] to give it a shot, really my first
time using asterisk. I've got my sipura 3000 working fine with
asterisk as an extension and I can dial out just fine, but incoming
calls to my toll-free DIDs are not working. I get the following
error:
May 28 04:57:47
http://asteriskathome.sourceforge.net/
--- Quintin [EMAIL PROTECTED] wrote:
Hi
Can any one tel me what is [EMAIL PROTECTED]
Thx
Q
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Hi all,whats the root password for [EMAIL PROTECTED] db, to
login from the consel?
Thx
Q
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Quintin wrote:
Hi all,whats the root password for [EMAIL PROTECTED] db, to login from the
consel?
Thx
Q
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From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Quintin
Sent: Thursday, May 26, 2005 1:15
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users]
[EMAIL PROTECTED] - mysql login
Hi all,whats the root password for [EMAIL
Hi
Can any one tel me what is [EMAIL PROTECTED]
Thx
Q
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Title: Message
Google
is your friend:
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2003-36,GGLD:enq=%22What+is+Asterisk%40Home%22
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
QuintinSent: 24 May 2005 11:41To:
Asterisk @ Home This CD will install
everything you need to get your Software PBX going.
Its a complete ISO CD that brings
together the OS (CentOS 3.4)
Asterisk Software version 1.0.7 stable
AMP Asterisk Management Portal
Web GUI
FTP
TFTP
Plus many more items.
Every
and man does it kick ass to!!! :)
email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn
messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5
- Original Message -
From:
Ariel
Batista
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, May
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
I am using Sipgate with [EMAIL PROTECTED] and this is how I have set mine up to
have it working perfectly. Using the AMP Interface my trunk is setup as
follows..
Under Trunk:
Outbound caller ID is your full sip number including area code.
Peer Detail:
allow=ulaw
authuser=539 (your sip
-- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack
-- Called sipgate/##
Paul I apreciate why you've the dialled digits out there, but
would you be good enough to include the first few, as if your asterisk
box is sending extra / unwanted / too few digits to
As far as I understand, it is in .gz file. Based on what schedule backup is
performed? Is it changeable? How to set more then one backup destination?
For example, over the net? How do I selectively restore files? For example,
I do want to restore only sip.conf and leave everything? Do I have to do
Hello,
How do I routinely backup all necessary configuration files on [EMAIL
PROTECTED] Is
there any procedure/tool/script for it? And if I need to move * with
existing configuration on a new hardware, what is the best way to do it?
Thanks
I.N.
___
@home do that for you everyday...;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irakli
Natsvlishvili
Sent: Sunday, May 15, 2005 2:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question
Hello,
I've got an Asterisk @home setup that mostly works. I'm using Polycom
SIP phones and have PSTN termination through an IAX provider. I say
mostly works because, if I'm on the phone, any additional incoming
call goes straight to voicemail rather than signaling call waiting on
the phone. I've
Howell
Sent: Wednesday, May 11, 2005 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk @home with IAX termination...
I've got an Asterisk @home setup that mostly works. I'm using Polycom
SIP phones and have PSTN termination through an IAX
From somewhere in Cyberspace to all
points on the compass its the On Computers Radio Show.
Sunday's show will be all tech talk with the
team. Peter Kastner will start off the show by talking about this week's hot
news stories then the gang will continue on with tech talk. During the
We installed AAH .05, tweaked it and learned more about dialplans, Queues
(not included in that version of AMP), upgrading to CVS Head (needed
atxfer/automon) and anything else we needed to scale AAH to our needs (75
agents 15k+ calls/day)than I believe we would have learned by simply trying
to
After installation of [EMAIL PROTECTED] v1, I have an annoying message
in the screen, anyone know how to fix it
INIT: Id s0 respawning too fast: disable for 5
minutes
Thanks
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This is not that AAH mailing list, check out the fourms.
On Sat, 30 Apr 2005, Manny A. Wise wrote:
After installation of [EMAIL PROTECTED] v1, I have an annoying message in the
screen, anyone know how to fix it
INIT: Id s0 respawning too fast: disable for 5 minutes
Thanks
I disabled serial port login by commenting out the ttys0 line in
/etc/inittab. Then after I rebooted the machine the message stopped.
There is a way to make this take effect without rebooting the machine,
but I was too lazy at the time...
I don't know what the real cause is though. If you
[EMAIL PROTECTED] 1.0 released
This is the first production release of [EMAIL PROTECTED]
We have worked hard over the past few months to make
[EMAIL PROTECTED] easy to use and stable. Thanks for all
the help with testing and fixes from [EMAIL PROTECTED]
users all over the world.
There are no
Hi
I am currently running [EMAIL PROTECTED] version 0.9 and have a few questions,
which i hope someone on this list might be able to answer.
1) I am trying to setup incomming fax support, but however i never manage
to receive the faxes, getting a signal 15. As per handbook, there isn't
too much
Title: Asterisk Home .9 with TDM11B
Hi all,
I am new to Linux so sorry for asking stupid questions. I have the TDM11B(TDM 400) card and I cant get it to work at all. Can someone point me to the right direction?
Thanks
___
Asterisk-Users
when you have asterisk install first log into the console and type
help-aah for some more info. Look also at the [EMAIL PROTECTED] handbook
online.
Then for the TDM11B card do a genzaptelconf -s -d.
Then you need to setup a zap trunk via your web interface. you might also
want to look at the
-- Executing Dial(SIP/3001-e13a, ZAP/1/65869804) in new stack
This is what's wrong I think. The line is missing the 'g' for the trunk
group. On all of my [EMAIL PROTECTED] boxes the cli shows
-- Executing Dial(SIP/227-a4dd, ZAP/g0/3428463) in new stack
It depends how you set it up
This is my sip trunk configuration
canreinvite=yes
context=from-internal
disallow=g729
host=192.168.1.138
mask=255.255.255.255
qualify=yes
type=peer
maybe it could be the context!
HTH
Edgar
On Thu, 2005-04-21 at 12:57 +0800, Dinesh wrote:
Great:)
Just one question, I am trying to get the
I was thinking about the best way to hook up the second line in my
house to an * fxs port. Would I just wire the fxs to the incoming
side of a line at my demarc? Or should I splice it in after that?
I need to rewire the whole house anyways. What I had imagined was new
cat3 for the
Chris,
If you are looking to run your second line through *, you will need to
run the line from the demarc to an FXO port on your Asterisk machine,
and then run a line from an FXS card to the jack location where you
will be using an analog phone.
You cannot connect an incoming line from your
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dylan
VanHerpen
Sent: Thursday, April 21, 2005 9:21 AM
To: snacktime; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk home wiring question
Chris,
If you are looking to run your second line through
Just to add some general wiring tips from my experience. (and yes you need
to go to an fxo port with the incoming telco line, and the fxs ports go to
your phones - you can't mix and match and parallel things up, the only
exception being more than one phone on an fxs port if the port can supply
Discussion
Subject: Re: [Asterisk-Users] asterisk home wiring question
Chris,
If you are looking to run your second line through *, you
will need to run the line from the demarc to an FXO port on
your Asterisk machine, and then run a line from an FXS card
to the jack location where you
Hi all,
Trying to upgrade to the new version, and all went well. I
am having some problems with the zaptel config.
When ever I try to dial a pstn number, I get this message.
I did a yum update and also recompiled the zaptel and asterisk modules in
/usr/src. I used the same machine
-- Executing Dial(SIP/3001-e13a, ZAP/1/65869804) in new stack
This is what's wrong I think. The line is missing the 'g' for the trunk
group. On all of my [EMAIL PROTECTED] boxes the cli shows
-- Executing Dial(SIP/227-a4dd, ZAP/g0/3428463) in new stack
I was thinking about the best way to hook up the second line in my
house to an * fxs port. Would I just wire the fxs to the incoming
side of a line at my demarc? Or should I splice it in after that?
I need to rewire the whole house anyways. What I had imagined was new
cat3 for the phones, and
Great:)
Just one question, I am trying to get the cisco callmanager with
[EMAIL PROTECTED] integration. Having some problem, if I edit the config files,
the [EMAIL PROTECTED] doesn't see the sip peers.
I am following this
Hi, does anybody ever tried using [EMAIL PROTECTED] with Digium
wcte11xp (E1) for R2 signaling? Im having problem compiling the libunicall
libraries.
Please help.
Thanks.
Angelo
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Hi list!
I'm just wondering if anyone has any experience with [EMAIL PROTECTED]
connecting to an ISDN BRI line, and if so, could you tell me what
hardware will work well with [EMAIL PROTECTED]
I've been looking at the Fritz!PCI but it seems (though
Frtiz is a nightmare although it is cheap and I have seen it
working. I have been trying to install it for some days without success but one
thing is for sure: you have to use the right Kernel (they are available for
2.4.20 and 2.6something). There are no clear instructions to install it,
Hi,
Il giorno sab, 16-04-2005 alle 13:30 +0200, Robson Ribeiro ha scritto:
Frtiz is a nightmare although it is cheap and I have seen it working.
I have been trying to install it for some days without success but one
thing is for sure: you have to use the right Kernel (they are
available for
On 13:30, Sat 16 Apr 05, Robson Ribeiro wrote:
Frtiz is a nightmare although it is cheap and I have seen it working. I have
been trying to install it for some days without success but one thing is for
sure: you have to use the right Kernel (they are available for 2.4.20 and
2.6.something).
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