Displaying 6 results from an estimated 6 matches for "leeb00".
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leeb
2004 Dec 10
2
[Fwd: Re: udev or not?]
Forwarded back to the list so others might get the benefit of the
answers, and I get fact checked by others.
-------- Forwarded Message --------
> From: Lee <leeb00@gmail.com>
> Reply-To: Lee <leeb00@gmail.com>
> To: Steven Critchfield <critch@basesys.com>
> Subject: Re: [Asterisk-Users] udev or not?
> Date: Fri, 10 Dec 2004 13:00:29 -0800
> On Fri, 10 Dec 2004 06:36:02 -0600, Steven Critchfield
> <critch@basesys.com> wro...
2004 Dec 11
5
does aanyone have an example of how to dial outwith a sip phone on a pstn line?
Charles S. Antrim wrote:
> I am using a card that has an fxo and fxs module.
I am no where near an expert but I have my sip phone working through my
pstn line and this is my config.
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext =
2004 Nov 30
3
Asterisk for home office
I apoligize in advance for this newbie question on what I perceive as
a mostly advanced level list... I did some searching, but would like
some of your expert opinions.
I'm building an asterisk server to be used in the home, both to learn,
and as proof of concept of applying this solution in a home. To keep
costs down, I'm considering one X100P ($25 ebay clone) card to connect
the
2004 Dec 10
1
udev or not?
I'm just getting started with asterisk on White Box Linux 3.0
I have a x100p card in the PC.
If I reboot and immediately do a "ztcfg -vv" I get the well known
"Unable to open master device '/dev/zap/ctl'. If I then do a "modprobe
wcfxo" this error disappears and Asterisk runs fine.
I've read the advice about modifying a file in /etc/udev. Well this
system
2005 Jan 01
0
Audio breakup problems
I've been having audio breakup problems (on my end) in my Asterisk
tests. I'm not sure of the most likely source of this quality problem.
99% of my LD calls are calling into a tele-conference service called
freeconference.com for group meetings. Its a free phone conference
system that works quite well with pstn phones. I've been using it for
quite some time. But the audio problems
2005 Jan 20
0
Asterisk@Home and iax.cc / sixTel
Hi,
How is iax.cc / sixTel to be configured as a termination provider in
asterisk@home?
The iax.cc / sixTel instructions tell you to do this:
iax.conf:
------------
[sixTel]
type = friend
host = iax2.sixtel.net
context = inbound
secret = mypassword
allow = all
extensions.conf:
--------------------------------
exten =>