Justin Sinclair wrote:
From: David Luyens [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] fedora core 1 install problem
Date: Mon, 22 Dec 2003 14:20:55 +0100
Reply-To: [EMAIL PROTECTED]
Hi Ernest,
I have installed as you described, and now it worked.
Seems that installing
I need to make automatic voice calls from a Linux server, so when the
system receive special signals it must use a wave (or .au) audio file,
dial the number to call a person, and speak using the audio file.
What can i use for this subject?
I need a specific hardware device or a normal analogic
hi,
i have been looking for any GUI that would make
things easier to configure friends and peers into asterisk. I also looked at
some posts in the lists. There are discussions that say text or CLI is more
appropriate for adding users and stuff. Anyone know of any interface that would
make
Davide Giunchi wrote:
I need to make automatic voice calls from a Linux server, so when the
system receive special signals it must use a wave (or .au) audio file,
dial the number to call a person, and speak using the audio file.
What can i use for this subject?
I need a specific hardware device or
Chandra wrote:
hi,
i have been looking for any GUI that would make things easier to
configure friends and peers into asterisk. I also looked at some posts
in the lists. There are discussions that say text or CLI is more
appropriate for adding users and stuff. Anyone know of any interface
Chandra,
Take a look at:
http://sourceforge.net/projects/astguiclient/
it may be what your looking for or you could use the ideas if you want to make changes.
I believe it was written by Matt Florell, Thanks Matt.
At 14:42 30/12/03 +0545, you wrote:
hi,
i have been looking for any GUI that
I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
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Title: Routing calls from a T1 based on DNSI.
Dear Group,
I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX.
On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality.
Does
I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html
You won't get the whopping 95 cent discount from BSD Mall but
is there a installation guide? i didn't find any.
just the readme file.
- Original Message -
From:
Peter Brown
To: [EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 2:58
PM
Subject: Re: [Asterisk-Users] Asterisk
Config thru web interface or any GUI
Hi !
I am trying to configure two E100P cards, but I am a bit confused with
zapta.conf in what I am trying to achieve.
The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines
will be used for incoming calls as well as outgoing calls.
My problem now is what to put in
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
All good here also.. this has got to be in the to 10 stupidest things
posted to the mailing list today.
Stupid as in development stopped, agreed.
However, there have been times in the past where cvs.digium.com has
Patrick Wong wrote:
Hi all,
I just checked out that Asterisk which is a platform I am interested
of. I would like to install it to the Linux box for a trial. I have
some legacy Dialogic hardware on hand, don't know they will work with
Asterisk or not. For analog loop start interface I
Steve Underwood wrote:
That card's haardware is not capable of providing any VoIP
functionality. It is not full duplex. The newer JCT cards can be used,
but they still don't work that well, due to card limitations.
This is exactly why people should support Digium. After all they have
GIVEN
On Tue, 2003-12-30 at 04:04, Shad Mortazavi wrote:
Dear Group,
I'm in the final phases of switching over from my existing PBX to an
Asterisk based PBX.
On my current PBX calls are routed on the existing PBX using a
assigned DNSI number, and I'm looking at replicating this
functionality.
Hi,
Anybody knows if Asterisk work fine with ser ?
We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN
termination for inbound and outbound calls.
Jorge
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Hi-
Not sure that I understand your question about grouping, but here is what I
use for 2 E1's connected to a private switch (in addition to the other
parameters) Note that I use the pri_cpe (customer premise equipment)
setting. The defined channels act as one big group of 60 channels, if
Carl,
You would have to know me to understand that was a JOKE! Most
people on the list and in the irc channel know already! Lighten up and
live a little.
Digium is about to setup CVS mirrors because if * is on /. one more time
and I have to do that .5kb/sec CVS checkout.. i'm gonna
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
Ahh, but the question is worded such that the virtualization is running
on windows. Therefore you have a lot of display overhead due to a
windows environment. You also are just an application running in an OS,
so
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Carl A. Cook
On Tuesday 30 December 2003 06:16 am, Adams, Gavin wrote:
Stupid as in development stopped, agreed.
Be advised, that the newest tarball is 4 months old. Can you explain
On Tue, 2003-12-30 at 10:57, Carl A. Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 29 December 2003 11:18 pm, Brian West wrote:
All good here also.. this has got to be in the to 10 stupidest things
posted to the mailing list today.
bkw
Very nice Brian, thanks
Don't say that.
Does that mean that from now on we will get a voice asking if we really want
to do that at every button press?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: 30 December 2003 18:27
To: [EMAIL PROTECTED]
Hi,
I read in the Asterisk Whitepaper, that you can run two cloned servers,
one as a primary, one as a backup, and have them automatically failover
to the other unit when it crashes, or when you need to restart it. The
primary application of course, would be ensuring calls can be made when
On Tue, 2003-12-30 at 12:34, David J Carter wrote:
Don't say that.
Does that mean that from now on we will get a voice asking if we really want
to do that at every button press?
heh, I can imagine it now, a call that says, A call has been received.
You will now need to restart your computer.
I simply have 2 asterisk servers and have the clients point to a DNS SVR
record for their proxy. The DNS record lists the primary and secondary
with preference for the primary. This won't stop calls from being
dropped if the primary goes down if you are routing them through the
server, but it
Hey,
I am currently working on a Polycom 500 phone Asterisk solution, and
the key is definitely to use the xml config files that Matt spoke of.
That combined with an FTP server (setup like the sip docs say) work very
well in getting the phone to do what you want. It then becomes getting
the
Title: Multi-line, multi-registration phones
I have hard phones that are capable of handling three calls at once. That is setup (apparently) through multiple registrations. My question is has anyone done this and what is the proper way of doing it? Do I have to setup (for 2 phones that have
What kind of channelbank/FXS port are you connecting to?
I've seen problems connecting to some of the older versions of the
Adtran Total Access 750's. I wouldn't doubt there would be problems on
other channelbanks with older firmwares. Of course, no firmware on CAC
AB1's
I have the AAstra
Title: X100p always busy - update
Well, after bummin around thinking I had bad fxo cards, I finally discovered that I was loosing two phones in my home when I had the cards plugged in Turns out the jack I had plugged into, was wired for two phones (line 1, green/red and line 2, yel/blk) and
Hi,
I've just read on the Wiki, that Asterisk can be compiled to run on Mac
OS X (BSD). How?!
I've just tried running 'make' through the command line, and it dies
from a gcc bug. According to the Wiki, it does state that there is a
specially modified make file that has been ported to
Brian West wrote:
its
#include filename.conf
Does the synatx include the # at the beginning of the line ?
And can this type of include be time/date dependant like the standard
include ?
include = filename.conf|hours|weekdays|monthdays|months
--
.~.
/V\Lance C. Arbuckle
On Tue, 2003-12-30 at 14:29, Sean Garland wrote:
I have hard phones that are capable of handling three calls at once.
That is setup (apparently) through multiple registrations. My
question is has anyone done this and what is the proper way of doing
it? Do I have to setup (for 2 phones that
Hi,
I have been trying to get my 7960 7960G to register with two seperate *
servers.
Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15
Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30
7960 is on the LAN running: P0S3-04-4-00
7960G is on
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lance Arbuckle
Sent: Tuesday, December 30, 2003 4:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] include a file ?
Brian West wrote:
its
#include filename.conf
Does the synatx
On Tue, 2003-12-30 at 15:16, Lance Arbuckle wrote:
Brian West wrote:
its
#include filename.conf
Does the synatx include the # at the beginning of the line ?
And can this type of include be time/date dependant like the standard
include ?
include =
No you guys need to pay attention..
include = context
#include filename.conf
They do totally diffren things.
bkw
On Tue, 30 Dec 2003, Lance Arbuckle wrote:
Brian West wrote:
its
#include filename.conf
Does the synatx include the # at the beginning of the line ?
And can this
heh, I can imagine it now, a call that says, A call has been received.
You will now need to restart your computer.
Or better yet... Who do you wanna call today?
bkw
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- Original Message -
From: Lance Arbuckle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 4:16 PM
Subject: Re: [Asterisk-Users] include a file ?
Brian West wrote:
its
#include filename.conf
Does the synatx include the # at the beginning of the line
Sean Cheesman wrote:
The # is needed. It's your standard programming syntax.
My two cents on the date/time variable would be no. The includes are
processed when * starts up, and are all grouped together. It's more of a
way to keep everything clean than for a logic basis. Anyone else?
Anton Yurchenko wrote:
Philipp von Klitzing wrote:
Hi!
Can such thing be done through dialplan , that say I transfer a call
to an extension but it is busy, so that this call returns back to me.
How about
- you store a temporary variable using SetVar() with the name of the
callerid
Steven Critchfield wrote:
On Mon, 2003-12-29 at 14:54, Olle E. Johansson wrote:
Steven Critchfield wrote:
[incoming]
exten = _.*,1,answer
;exten = _.*,2,agi(timeout-lookup.agi) ; alternative
exten = _.*/some match,2,Absolutetimeout(360)
exten = _.*,2,noop
exten =
include = context is for context inclusion
#include filename.conf is for including other files that follow the standard
config files (sip, extensions, etc)
Don't let the two includes confuse you. They serve two completely different
functions on two completely different levels. If you had 1000
Philipp von Klitzing wrote:
Tried to DL using CVS this eve, and it says:
Unknown host cvs.digium.com.
Has Asterisk development stopped?
Digium was just sold to Microsoft.
I must reset my date, is it already april 1:st?
exten= 20040401,1,SayKlitzingTime()
/O :-)
On Tue, 2003-12-30 at 16:25, Olle E. Johansson wrote:
OK, thanks.
Wasn't aware of that. Interesting feature.
I'll see if I'm able to document it so users understand. It opens for very
unreadable configurations, so, readers, please use with care... :-)
Just like in programming, anytime
Steven Critchfield wrote:
On Tue, 2003-12-30 at 12:34, David J Carter wrote:
Don't say that.
Does that mean that from now on we will get a voice asking if we really want
to do that at every button press?
heh, I can imagine it now, a call that says, A call has been received.
You will now need
Steve Dolloff wrote:
I simply have 2 asterisk servers and have the clients point to a DNS SVR
record for their proxy. The DNS record lists the primary and secondary
with preference for the primary. This won't stop calls from being
dropped if the primary goes down if you are routing them through
Thanks for all your help Martin,
Guys,
This is a good find and hopefully could help someone else.
I've been having a problem with forwarding voicemail from one mailbox to
another. I ran down the sendmail and soundcard path and came up goose eggs.
With intuitive guidance from Martin Pycko
Lance,
Parsing of configuration files is done at CLI reload or startup. That includes the
#include *FILE*
construct.
The include statement - without the # character - includes *contexts* and this can be
done
at different times, since all contexts are parsed when Asterisk parses configuration
(new asterisk user - currently setting up Polycom IP600 phones)
Does anyone know if it's possible to make a sip phone instantly pick up
on speakerphone when a particular call comes in? Eg so that you can
quickly bother someone across the office without making them reach for
their phone?
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the please hold while I try that extension
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten = 8005,1,Macro(stdexten,8005,Zap/2)
exten =
Hello All,
I managed to Configure TDM400P, now I can call Analog Phones connected to TDM400P from
SIP Phones (CISCO and SNOM).
But when I try to dial any number from Analog Phones I get following message
-- Starting simple switch on 'Zap/3-1'
-- Hungup 'Zap/3-1'
My extension.conf has
Did you try with this line before launching asterisk (with stock redhat
9 kernels):
export LD_ASSUME_KERNEL=2.4.1
Best regards,
On Tue, 2003-12-30 at 20:07, JR Richardson wrote:
Thanks for all your help Martin,
Guys,
This is a good find and hopefully could help someone else.
I've been
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Here is an example of a couple of macros that help me where I have a SOHO with a
home phone line and a work phone line. If I pick up line 2 my work line I would
prefer the call I make to go out my office phone line same with if I pick up
line 1 my home phone line I would prefer it go out my home
RedHat 9 and Fedora kernels have a new feature (not present in
kernel.org): Native Posix Threads
This brings all sort of problems to diferente applications. To override
this new feature, you have to start your affected programs with that
enviroment variable set.
On Tue, 2003-12-30 at 21:43, JR
Sure, head to :
http://store.yahoo.com/asteriskpbx/wildcardx100p.html
-d
At 12:25 PM 12/30/2003, you wrote:
I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
- Original Message -
From: Patrick Wong [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 9:20 PM
Subject: Re: [Asterisk-Users] Does Asterisk support legacy Dialogic
products?
Steve Underwood wrote:
That card's haardware is not capable of providing any
Hi,
I move the * on a new DELL server and I get the latest version of Asterisk
with CVS. I have 3 FXO cards, X100P and the sound before was fine.
With the new version of Asterisk and on new Dell server the sound is
SO BAD!
Some suggestions are welcome.
Best regards,
Chris HARIGA
Update #2
According to the helpful folks at Sayson, this phone actually has two
'slots', and the ADSI stuff needs to be downloaded to both (the second one
kicks in when the phone is idle for more than 1s). I had only downloaded
asterisk.adsi to the first slot, which explains why I was seeing the
On Thu, 18 Dec 2003, Aaron Martin wrote:
I have upgraded my grandstream phone from firmware 1.0.3.78 to
10.0.4.30 and now I am having problems with early dial. On the older
firmware earlydial worked fine with my asterisk server, but now as soon
as I have dialed the number I get a
On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote:
On Thu, 18 Dec 2003, Aaron Martin wrote:
I have upgraded my grandstream phone from firmware 1.0.3.78 to
10.0.4.30 and now I am having problems with early dial. On the
older firmware earlydial worked fine with my asterisk server, but
On Thu, 18 Dec 2003, Aaron Martin wrote:
I have upgraded my grandstream phone from firmware 1.0.3.78 to
10.0.4.30 and now I am having problems with early dial. On the older
firmware earlydial worked fine with my asterisk server, but now as
soon as I have dialed the number I get a congested
I'm trying to use accountcodes, but experiencing inconsistant
results. I have two * servers, one which appears to be working as
expected and one not. I would like to prepend the device's accountcode
to the dialed number. The sip1 server does not seem to have the
${ACCOUNTCODE}
hi all
i need constulancy service for depolying Asterik
below are my requirement in point forms
All Single Line telephone are attached to an
autodialer which is program to dial the access number
and store digits dial by the user.
Ie. When the user picks up the phone and dials
Hello,
I have been retained by a Building Management Company to install a
combined Voice/Data solution for a Tennated Office Space. This space will
rent offices, with telephone and internet service to inviduals or small
groups of individuals. As fate would have it, the service will be
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