Displaying 16 results from an estimated 16 matches for "goliv".
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golin
2006 Mar 02
2
Help with routing error.
...ttp://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<meta http-equiv="content-type" content="text/html;charset=utf-8" />
<meta name="generator" content="Adobe GoLive" />
<title>Form | Contact A</title>
</head>
<body bgcolor="#ffffff">
<form action="cgi-bin/parse.rb" method="post" #I have tried get
enctype="application/x-www-form-urlencoded"> #I have tried text/plain...
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....
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
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2003 Apr 15
0
FW: Academic Software Discounts
...ss.
---------------------- Education Standard You
ADOBE (Windows & Mac): Price Retail Save!
---------------------- --------- ------ -----
Acrobat 5.0 $57.95 $249 77%
After Effects 5.5 $289.95 $649 55%
GoLive 6.0/LiveMotion 2.0 $84.95 $399 79%
Illustrator 10.0 $89.95 $399 77%
InDesign 2.0 $189.95 $699 73%
PageMaker 7.0 $279.95 $499 44%
PageMaker 7.0 Upgrade $89.95 - -
Photoshop 7.0...
2005 Sep 21
0
Recommendation for HTML editor (a better approach?)
...t; > 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 utility that, like GoLive, will track
> > > which documents have been modified since last upload and send only
> > > those?
> >
> > Why not use find and -mtime to get a list of files that were modified
> > and then FTP that?
> >
> > If you combined that with a "touch&q...
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 is: What knowledge must I have to
able to understand what's go...
2006 Mar 05
5
lighttpd / WEBrick forum?
.../xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<meta http-equiv="content-type" content="text/html;charset=utf-8" />
<meta name="generator" content="Adobe GoLive" />
<title>Form | Contact A</title>
</head>
<body bgcolor="#ffffff">
<form action="cgi-bin/parse.rb" method="post" #I have tried get
enctype="application/x-www...
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.]--
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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.
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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