Displaying 16 results from an estimated 16 matches for "goliver".
Did you mean:
oliver
2006 Mar 02
2
Help with routing error.
Ok .. if you are sensitive ... stop reading.
I am totally new to setting up CGI. I am getting a routing error.
Would someone be kind enough to help me? Please?
Here is my form ...
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
2005 Jan 05
1
hosts and interfaces
interfaces:
local eth0 192.168.1.255 dhcp
golive eth1 172.30.15.255
wiredc eth2 202.37.230.127 dhcp
wave eth3 203.96.213.255
hosts:
ipsec eth2:192.168.192.0/24
rules:
DNAT wiredc local:192.168.1.3 tcp 80 -
DNAT wave local:192.168.1.3 tcp 80 -
203.96.213.73
The rules here
2006 Mar 08
5
Cisco 7960 SIP - Displaying Time
Is there a way to display the time of the 7960 running firmware 7.4? Im
unable to find any information.
Thanks,
Ben Blakely
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm
2003 Apr 15
0
FW: Academic Software Discounts
Macromedia Studio MX at 76% OFF,
Adobe Photoshop at 56% OFF,
Office XP Standard at 71% OFF,
Adobe Design Collection at 62% OFF
Microsoft Visual Studio.NET at 91% OFF,
FREE SHIPPING THROUGH APRIL 30, 2003
WITH BELOW CODE.
FREE SHIPPING CODE (Ground Only): DKG402
YOU MUST PLACE ORDER BY TELEPHONE AND
YOU MUST TELL OUR OPERATOR THAT YOU
HAVE A FREE SHIPPING CODE.
Dear Students, Teachers,
2005 Sep 21
0
Recommendation for HTML editor (a better approach?)
From: Scot L. Harris [mailto:webid at cfl.rr.com]
>
> On Tue, 2005-09-20 at 23:45, Greg Knaddison wrote:
> > On 9/20/05, Dave Gutteridge <dave at tokyocomedy.com> wrote:
> > > Which is my long winded way of coming around to asking - what would be
> > > the best approach to uploading data from my machine in this
> > > circumstance? Is there an FTP
2005 Jan 04
0
Dell Poweredge 6300 & 4 analogue lines
I'm just about to start implementing this project. I have a test
server working well with SIP phones and IAX for incoming and outgoing,
but when I golive will need 4 analogue lines coming in.
1. Anyone got this config working with a 4 port FXO digium card
2. Any tips/hints/traps
Thanks
John
(My first posting be gentle)
2001 Sep 01
0
adobe apps installed finally but
They don't work. With Codeweavers preview 4 I just had to place a few
native dlls (easy the program asked for them) and I managed to install
Photoshop5.5, GoLive5, and Illustrator9. But Photoshop won't save
files, Illustrators pentool is totally out of control, and GoLive will
freeze on start up. So now what? Do I have to learn C if I want to
progress from here? My real question
2006 Mar 05
5
lighttpd / WEBrick forum?
Is there a mail list or USENET group for lighttpd WEBrick questions.
I posted a simple CGI question to this list last week and got one
responce and that responce told me I was on the wrong list with Ruby
based from / CGI questions.
The code I posted looks correct but does not work giving a Routing
Error. This makes me think something is wrong with the set up of
WEBrick. I am only doing
2003 Mar 03
40
callerid
"In general you can match callerID with the /, but if you don't put
anything after the /, then the rule matches "no caller*ID", and if no
slash is there at all, it matches "any callerid". "
Ok.My question is ->
how to match callerid from 001... ?
and if don't know how many numbers ?
exten => s/0_,Answer don't work-
anything else ?
tnx
Thomas
2003 Aug 21
20
Your details
--[This is an automatically generated email notification.]--
************************************************************
M A I L W A T C H V I R U S A L E R T !
************************************************************
You recently sent a message containing a known virus -
W32/Sobig.f@MM (ED).
The message was sent on 8/20/2003 11:39:38 PM.
The subject of the message was:
Your
2006 Mar 03
9
Program Buttons on Cisco 79xx Phones
Does anyone have a good resource to learn how to program the soft and
hard buttons on a Cisco 7940 or 7960 phone? Using SIP
Firmware...thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm
2004 Jul 27
6
Asterisk to CCM
I've got problem with connecting asterisk to CCM.
Our side has Asterisk system other side CCM , ehrn i dial a number on
other side channles created , connections established but nothing happend
, just silence , and after some time busy tone. Sides sending ad reciving
g711 codec , but we need that sides send and recive g729 (we have
licenses) , if in h323 conf i try to : disallow=all ,
2006 May 22
2
I've broken voicemail
I went to put in the new sound files over the weekend, but forgot to backup
the custom folder and lost my custom digital receptionist files.
I then had to copy the old files back from a duplicate machine.
The problem is now though that voicemail just hangs up when I dial it.
Other apps work - *70 for example gives me 'call waiting...activated' so I
know it's accessing the files
2005 Feb 15
1
Asterisk "no one is available to take your call"
OK - I can successfully make calls from SIp phone through an asterisk
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
The problem is that if the call is not answered within ~5 seconds, *
gives the message "no one is available to take your call" and
disconnects the call. If I answer b4 the 5 seconds - everything is good.
Anywhere I need to set to get around
2005 Jan 24
0
Asterisk v1.0.1 Cisco 7960 Sip v7.3
Running those versions of code, my 7960 will not register with Asterisk.
The same 7960 is authenticating against another * server on line 2 just
fine though - with the same settings in sip.conf. On the failing *
server I am just getting 401 unauthorized errors on the console. From
the phone's shell I get that t is registering, but not authenticated .1.
from show reg.
Any ideas would be
2005 Feb 16
0
Outbound calling timeout
I am running asterisk 1.0.1 with OH323 compiled in.
We have a 323 trunk to CallManager with a mgcp controlled pri router.
When using sip phones (directly registered with asterisk) to call out
the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3
rings - no problem, otherwise I get "no one is available to answer at
this time" on the consoel and it redirects to an