On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote:
I don't see any major changes in the release notes--mostly small bug
fixes. They fixed some DHCP and NTP problems, as well as a 802.1x
problem with some of their switches. There were a couple SIP protocol
fixes in
On Sun, 27 Mar 2005 20:06:39 +1000, Chris Lee [EMAIL PROTECTED] wrote:
Ie: The phone doesn't appear to be grabbing the date time off the
NTP server on my network, it worked alright on 7.3 (except for the
time drift) but now they seem to have fixed the drift by no longer
displaying time nor
Colin Anderson wrote:
The hack came in through ssh.
IMO, your best defence is an extremely strong root password; I am often
mortified by looking at my logs and seeing all of the login attempts through
SSH.
OT: I am not up on Linux script-kiddie type tools, but I assume that there
is a script of
DEMAINE Benoit-Pierre wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
~ I would like to : set up a server on Linux on which my friends can
connect with msn or netmeeting, suporting at least sound conferance, and
optionally video, but I dont want asterisk server to lock up the sound
card; and
Joe Antkowiak wrote:
There are quite a number of positive (for end users) implications of
doing this too... just think about all those cell providers that
offer unlimited mobile to mobile calls, and then all those unlimited
LD packages from landline and voip providers. This has huge potential
[EMAIL PROTECTED] wrote:
Hello!
I would like to call a number (e.g.35), and when i press a secret code
(12345), it should jump to my voicebox menu.
On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found
something about Silently Wait for DTMF Input.
In my case it wouldn`t be
Steve Underwood wrote:
Chris Lee wrote:
I am looking at building an IVR product with a few interesting
features and need some more information about how asterisk and VoIP
work and what I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
information travels
Jon Radel wrote:
Marconi Rivello wrote:
In US, local calls are free. So it wouldn't be a problem to make such
a network to get rid of long distance calls. But in other countries
(like here in Brazil) local calls are charged. So there could be some
king of billing (without commercial purposes, just
I am looking at building an IVR product with a few interesting features
and need some more information about how asterisk and VoIP work and what
I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
information travels as data representing 'start tone' and
Miroslav Nachev wrote:
Hi,
I am looking for GSM to BRI ISDN Gateway. Any help?
I was also looking for such things nd came across these guys:
http://www.2n.cz/export
they have a product or two for GSM
and here is the one I found most likely to work for me (two GSM sim
cards providing two ISDN
James Freire wrote:
Hi All,
I have a server setup with an incomming PSTN line and a bunch of
Grandstream BT100 phones. Is there a way for asterisk to foward an
incomming callerID from the PSTN to the SIP phone that is setup as an
extension? We have a Voice menu setup for incomming calls and I
[EMAIL PROTECTED] wrote:
Hello List!
I just read an article about asterisk, and i would like to ask a few
questions to see if i understood the principle right.
Reciving Calls:
---
- To be able to recive calls, i need to have an VoIP-Provider.
No, You can use one but you could also
the problem maybe I can use asterisk and prevent audio
channel DTMF collection (is this simple) but if it is elsewhere what can
I do? (change handsets?)
Thanks for any insight on this
Regards
Chris Lee
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get echo issues? If not could you let us have your config and
which echo canceller you use.
Thanks
Chris
On Thu, 2004-06-24 at 20:40, Chris Lee wrote:
Chris Stenton wrote:
I am finding that I have to increase the txgain in zapata.conf to 8 when my
X101P is connected to my BT phone line, otherwise
Chris Stenton wrote:
I am finding that I have to increase the txgain in zapata.conf to 8 when my
X101P is connected to my BT phone line, otherwise people can hardly hear me.
This then gives echo issues.
Do other people have the same problem on BT lines. I was wondering if I
should give BT a call
B. J. Bomar wrote:
Hello all. We have an asterisk system set up, and we are seeing a lot
of multiple DTMF digits being read by asterisk. In digging through the
archives the only answer I have seen is to put in the statement
relaxdtmf=yes in the zapata.conf file. Since we are not using any
SNIP
On the other hand... Go take a look at all of the ~$100 wireless
router/firewall/print server/gateway boxes on the market, and you'll see
one thing that almost all of them have in common: they all run Linux.
Most of them are even based on the same small number of tools; things
like
Michael Hamann wrote:
Hi Everybody,
as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.
The first thing ist when I do a
Kevin Walsh wrote:
Steven Critchfield [EMAIL PROTECTED] wrote:
You forgot to add in how awful it is when people post using HTML and
then override font sizes or assume blue is an appropriate font color for
their message.
While I know some people don't like it when I turn my attention to them,
if
usedcanon wrote:
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
The London code is 020 the 7 or the 8 is part of the local number now.
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Hi
I am in the UK and am looking for a device that will allow me to connect
two sim cards (read wireless lines) to either the port on the back of my
fritz card or any other connection direct to the PC that provides a
usable telephony interface.
I will even plug two devices into a windows box
Storer, Darren wrote:
Hi Chris,
CL All I want is two GSM lines that look like voice modems to
CL the PC and provide full telephony interface, that is DTMF
CL both ways CLI and a few other bits and pieces.
We use the Nokia 22:
http://www.nokia.com/nokia/0,,56024,00.html
They have worked well
Storer, Darren wrote:
Hi Chris,
CL Does the incoming DTMF and voice work over the serial
CL interface with the 22?
I can't help but feel that you are going about this all the wrong way (based
upon the limited information you have chosen to share with us). If you need
to pass control information
that a key is held down is important so need the SIP device
to be sent the start and end of each key press the user makes.
Regards
Chris Lee
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Joost Kraaijeveld wrote:
Hi all,
I want to see the name of the caller (if available) and not the number.
If I call from my IP phone to my software IP phone I see the name of the caller. If I call from the software phone to my IP phone I only see the number, not the name. If I call from IP phone
The UK is currently not legally set up to allow the use of these devices
as there is currently no unlicensed bandwidth in the GSM space.
If you want to use one you have to approach the Cellular networks and
ask them to install it and connect it directly to their network.
You may be able to pay
Brian Cuthie wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roderick Montgomery
Sent: Monday, March 29, 2004 4:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
...
###
### Hardware != Software
###
Cisco IOS
First, is the lack of UK CLI on the x100P hardware or software related?
Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK
CLI and the same functionality as the x100p using a USR Modem with *?
Has anyone done this?
As an aside, has anyone experienced or solved the problem
Tim Sailer wrote:
On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote:
Simon,
Do the GS phones support stutter tone as-well-as
the message light?
I'm not Simon, but yes, they do. At least my -100 does. The display
backlight flashes, and you get the stutter dialtone.
Tim
My
=no
extensions.conf
[sip]
;local extensions
exten = 2000,1,Dial(SIP/2000,20,rt)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
voicemail.conf
[local]
2000 = 1999,simon,[EMAIL PROTECTED]
Hope that helps
Simon
Chris Lee wrote:
Tim Sailer wrote:
On Mon, Mar 08, 2004
When handed a URL type address for telephony, is there a DNS TX record
(like MX but for telephone/Video) that could be looked up for an address
to use to connect the call?
I would like to have a gateway server (probably *) that anyone who
knows the email address of a member of staff can use to
I am not a programmer so can not implement this, but I think it may be
useful.
Asterisk configured to listen on multiple IP addresses,
Then configure RTP ports for each address independently;
So I open 5 ports on one IP and then forward those ports to that IP
from my firewall.
Then on another
Thomas M. Schaefer wrote:
Hi all, I have a strange problem. Whenever I plug in the base cord connected
to the X100, my DSL service goes down. I DO have a Cisco filter (the one
that comes with the product) installed.
Has anyone else seen this problem?
There was a similar entry in the archives, but
I have been trying to set up three * servers to use IAX between them and
am a bit lost as to the finer detail of the config files. I have read
the wiki and it has not made things better.
Here is my problem;
I create a section like this on each machines:
[othermachine-1]
type=friend
Is there a howto for the situation below?
* -- router with nat port forward to * -- router with nat port
forward to sip phone -- sip phone
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To
Wes Marderness wrote:
What does your extensions.conf look like? Did you answer() the call first ?
The relevent sections of extensions.conf:
[voicemail access]
;Extension 8 to get to voicmail:
exten = 8,1,Answer
exten = 8,2,VoicemailMain
[wellingborough-road]
;includes
include = emergency
include
place for naming phone set to p3000 and p3001
place for port set to 5060
Thank you for any help
Regards
Chris Lee
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help would be great
Regards
Chris Lee
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On the subject of South Africa
What are the laws regarding using the Internet to carry telephone traffic?
What are the laws regarding connecting digium kit to Telkom equipment?
As I recall they are quite restrictive, have they been eased up a bit?
Regards
Chris
I followed the wiki instructions:
http://www.voip-info.org/wiki-Asterisk+non-root
Now I have a working asterisk running as user asterisk.
I do however have some problems:
1: I dont have access via asterisk -r
2: The pid file is no longer being updated
3: I want to create a file in init.d so that
Tilghman Lesher wrote:
Permissions problem. User asterisk needs to have permissions to
write the file /var/run/asterisk.ctl
2: The pid file is no longer being updated
Again, permissions problem.
I was under the impression that changing the line:
I am having problems with my dial plan, please help me locate the problem:
In the following dialplan, I am not able to press 8 to get to voicemail
main while the 3000 mailbox unavailable message is being read in the
background.
What am I doing wrong?
[globals]
;physical-phones
p1 = SIP/p3000
Bob Klepfer wrote:
voicemail is misspelled - would that do it?
Yup that fixed it, thanks for all the help
Regards
Chris
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not installed something or is there a
config file which has a new format as of 0.7.1?
please help
Regards
Chris Lee
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Building * on a machine with a minimal install of Mandrake, worked fine
on non minimal install now I get this:
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
If anyone can help me figure out what package I might have missed out
when installing mandrake,
I am looking to produce a switchboard interface - hopefully web based
I needs to:
Show the logged in user the CLI of the call they are currently dealing with
Show the number of calls in the queue
Give a number of options for working with the call
transfer
put on hold
etc.
for transfer it
As a sugestion, store the sounds in a soundlib tree, hashed or
categorised (boolean (yes, no, true,false, up, down etc.),numbers,
caledar(day, date, time etc), state, weather etc) and dont duplicate any
sounds then make a sounds tree with virtual categories and sim link to
the files needed.
I am new to asterisk and am wanting to know if it can do some things:
in a large/ distributed environment users move about either office to
office or branch to branch can they log in and have their virtual
extension routed to the one they are on?
naturaly this implies the question: if branch
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