Cygwin does have the cvs command, but it isn't installed as default. Go back
and tick more option when doing your install.
Tan
- Original Message -
From: "it" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 02, 2003 5:41 PM
Subject: Re: [Asterisk-Users] How could I get
I have setup a couple of openmosix clusters, not commercially just for interest sake..
I don't think the Voice/PBX type application would be suited to this type of
clustering because the voice system requires as close to realtime response as
possible.. The cluster works by load balancing the var
Hi.
I'm experiencing that bug with flash on zaptel.
That's the problem:
Zap/A call Zap/B
Zap/B flash transfers to Zap/C
Now Zap/A is online with Zap/C
Till now all ok...
but now if Zap/C wants to transfer again,
it can't... the debug says that it got a
WinkFlash when call not up or ringing
(as a
I tougth about using lvs and balancing incoming traffing.. but never
mosix..
On Wed, 2 Apr 2003, Liaan van der Merwe wrote:
> Hallo all
> Just for interest sakes..
> Have any of you tried to run * on a openmosix cluster?? Could be nice for
> ppl like me that dont have the resources to buy prop
Hallo all
Just for interest sakes..
Have any of you tried to run * on a
openmosix cluster?? Could be nice for ppl like me that don’t have the resources
to buy “proper” machines (I can get P1 166 for almost nothing, but higher than
that is to expensive)
Thanks
Regards,
Liaan W van der
Hi Jeff..
What you are asking is a little bit of a grey area because there are a number of
factors that will affect how well you system will perform.. things like the average
number of concurrent calls?, are you using VoIP?, what codecs are you using for the
SIP of IAX channels? and no doubt a
> But if I try to call from one of them to the other, the remote end rings
> just fine in both cases, but then as soon as asterisk bridges the two
> channels, the remote end sends a "Call/Leg Transaction Doesn't Exist"
> error and hangs up the line.
Apparently it doesn't like our reinvites for som
I know I've seen this reported already, and I can't remember the fix.
I have two ATA186s talking to an asterisk server. When I call in on an
outside line, both ring, and I can pick up either and talk.
But if I try to call from one of them to the other, the remote end rings
just fine in both ca
Hi folks,
Right now I'm running * along with a lot of other apps on my firewall box,
which is a P-II 400 with 192MB of RAM. I have a single T100P card connected
to a channel bank that's using one FXO and two FXS ports.
I want to move * off to another computer (mostly because I think the other
Hey do we have the ability to incriment a variable?
exten => t,2,SetVar,looptest=$((looptest + 1))
I was thinking of doing a library of simple arithmetic and bash-like
expansions for asterisk... like Zap/{1&,2&,3} - but it may already have this
functionality.
John
> exten => t,1,GotoIf(${looptest
Thank you very much. I am using the Windows Binary CVS now.
- Original Message -
From: "Chris Albertson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Tuesday, April 01, 2003 5:46 PM
Subject: Re: [Asterisk-Users] How could I get * from CVS if I am not on the
Lin
There are MANY CVS clients that run under non-linix.
Here are even native Windows point and click clients
and some java clients. Some work with web browsers and
don't even require that you install software on your
computer.
Look here and read the FAQ and browse the download section
for a Windows
there is also WinCVS. Nice GUI, too.
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 02, 2003 11:12 AM
Subject: Re: [Asterisk-Users] How could I get * from CVS if I am not on the
Linux platform?
> then download the sources an
How can I determine what kind of signaling I have. Tallhassee uses a Nortel DMS-100.
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 01, 2003 7:51 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Line is stuck off hook...
On Tue, 2003-04-0
A while ago SIP transfer via the # key on a call to a cell phone via
iconnect was working. I updated to the current CVS tonight and now that
functionality is gone. Any ideas as to how to enable it again?
Thanks in advance
-russ
___
Asterisk-Users mai
then download the sources and compile it ...
On Wed, 2 Apr 2003, it wrote:
> I installed the cygwin yesterday. But it seems that the cygwin does not have
> the cvs command.
>
> $ cvs
> bash: cvs: command not found
>
>
> Regards
> john
>
>
> - Original Message -
> From: "Michael Bielicki"
On Tue, 2003-04-01 at 18:38, Gene Kochanowsky wrote:
> How long is 600 in seconds?
I hope that is a April fools joke.
60 seconds to a minute, 600/60=10 minutes. 10 minutes is 1/6th of an
hour.
> -Original Message-
> From: Tilghman Lesher [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, April
Make sure you're using fxs_ks signalling for the FXO channels and also
make sure that your incoming lines support disconnect supervision.
Otherwise, * has no idea when the calling party hung up.
> Hi Steven,
>
> I have analog lines connected to the fxo lines of the Zhone channel
> bank. All of y
On Tue, 2003-04-01 at 18:29, Gene Kochanowsky wrote:
> Hi Steven,
>
> I have analog lines connected to the fxo lines of the Zhone channel
> bank. All of your suggestions sound good. How do you set up the config
> file so it would play the greeting twice and then hangup the line?
In your timeout e
How long is 600 in seconds?
-Original Message-
From: Tilghman Lesher [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 01, 2003 7:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Line is stuck off hook...
On Tuesday 01 April 2003 17:13, Gene Kochanowsky wrote:
> Greetings,
>
> I am
I installed the cygwin yesterday. But it seems that the cygwin does not have
the cvs command.
$ cvs
bash: cvs: command not found
Regards
john
- Original Message -
From: "Michael Bielicki" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "duncan" <[EMAIL PROTECTED]>
Sent: Friday, March 28,
Hi Steven,
I have analog lines connected to the fxo lines of the Zhone channel bank. All of your
suggestions sound good. How do you set up the config file so it would play the
greeting twice and then hangup the line?
That would fix the problem for the most part, but why isn't * releasing the li
It might have something to do with the zhone. I posted this on the 'Serious
problem with z-plex 10 thread' from this morning - see if any of this is
familiar,
I have observed that if you dial a (unused) port that does not have a phone
plugged into it it may start detecting pickups on unused ports a
On Monday 03 February 2003 11:37, Michael Manousos wrote:
> Hemant Kumar wrote:
> > inline...
> >
> > Rattana BIV wrote:
> >> Hi,
> >>
> >> There\'s anybody who knows how can I use h323 with Asterisk.
> >> I install pwlib 1.3.11 and openh323 1.9.10 but I have warning
> >> message and i can\'t lauch
On Tuesday 01 April 2003 17:13, Gene Kochanowsky wrote:
> Greetings,
>
> I am running Asterisk with a T100P and a Zhone channel bank
> for over a month now. For the most part it works fine but from
> time to time (about once a week) the system will not let go of
> a line and will play the greeting
On Tue, 2003-04-01 at 17:13, Gene Kochanowsky wrote:
> Greetings,
>
> I am running Asterisk with a T100P and a Zhone channel bank for over a
> month now. For the most part it works fine but from time to time
> (about once a week) the system will not let go of a line and will play
> the greeting ov
Greetings,
I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For
the most part it works fine but from time to time (about once a week) the system will
not let go of a line and will play the greeting over and over. Anyone calling gets a
busy signal. If I reset Ast
On Tue, 2003-04-01 at 15:44, WipeOut . wrote:
> > At my office, I have a context [outside] that defines the local and long
> > distance calling dial pattern matches. All the phones in my office are
> > dropped in the [office] context which is able to include the [outside]
> > context. The incoming
> At my office, I have a context [outside] that defines the local and long
> distance calling dial pattern matches. All the phones in my office are
> dropped in the [office] context which is able to include the [outside]
> context. The incoming lines are dropped in the [default] context. The
> [def
On Tue, 2003-04-01 at 15:08, sjaak nabuurs wrote:
> Thanks Steven for your reply
>
> To check if i understand this..
>
>
> So the best way is :
>
> Telephone line >> pluged in one TDM10B Linux running astrisk
> (DSL2Mbit) internet (64Kb) linux/windows with a normal eth 10/100mbps
>
Thank you. It is working now.
Allan
--- Martin Pycko <[EMAIL PROTECTED]> wrote:
> Read www.digium.com at Documentation->FAQ
>
> Martin
>
> On Mon, 31 Mar 2003, Allan Wang wrote:
>
> > Steven,
> >
> >
> > Could you please give me your config files for
> X100P
> > and S100U? I just got mine
Thanks Steven for your reply
To check if i understand this..
So the best way is :
Telephone line >> pluged in one TDM10B Linux running astrisk
(DSL2Mbit) internet (64Kb) linux/windows with a normal eth 10/100mbps
card pluged in >>> snom 100
When you talk about ethernet jack you tell
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of WipeOut .
> Sent: Tuesday, April 01, 2003 4:29 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Up to 8 lines?
>
> Hi,
>
> I am starting to look at the costs involved in putting
Title: CDR Format and Software
Greetings,
I am looking for a good CDR reporting tool to use with Asterisk.
Does anyone have a suggestion?
Additionally, is the CDR format that Asterisk uses the same as any commercial units on the market?
Most of the software I have looked at so far has a
On Tue, 2003-04-01 at 13:43, Only Yours wrote:
> Hi guyz
>
> I m very much new to Asterisk and i want to establish
> a PABX for my house at the moment i have
>
> X100 card(Configured)
> Asterisk PABX(Installed)
>
> Now i want to know how would i recieve call for e.g.
> i want during incoming ca
Hi guyz
I m very much new to Asterisk and i want to establish
a PABX for my house at the moment i have
X100 card(Configured)
Asterisk PABX(Installed)
Now i want to know how would i recieve call for e.g.
i want during incoming call (after demo mess). if user
press 9 (for e.g) the call redirect t
Hi,
I picked up a router with 8 voice ports that supports MGCP, but it has
several options that I am not familiar with or do not seem apparent in the
mgcp.conf.
Enter the default IP address for the Notified Entity: [0.0.0.0]
Enter the listening port of the Notified Entity: [2427]
Enter the IP add
E&M winkstart, loopstart and groundstart signalling should work without
problems. We don't support Qsig signalling yet.
regards
Martin
On Tue, 1 Apr 2003, Eduardo Goncalves wrote:
> I can work with digital E&M - winkstart, immediate, loopstart, groundstart and
> ISDN with Qsig. Also with
William, did you ever hear back from ActionTec?
- Original Message -
From: "William X Walsh" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February 26, 2003 12:11 AM
Subject: RE: [Asterisk-Users] ActionTec with Asterisk
> I'm in touch with their business development team,
On 1 Apr 2003, Steven Critchfield wrote:
> On Tue, 2003-04-01 at 08:14, Ahmed Boreau wrote:
> > hi,
> >
> > I'm trying to install asterisk server on mandrake 8.0 and I got a
> > dependencies problem with libreadline.so.3
> >
> > I downloaded readline-4 and readline-2.2.1 package and it still
On Tuesday 01 April 2003 04:54, Stefano Finetti wrote:
> - If I call a toll-free number with an IVR system, nothing
> happens: it continues to ring indefinitely like no-one answers
> the call.
>
> This is my part of extensions.conf related:
>
> ; Calling Mobile Numbers
> exten => _3N,1,SetC
On Tue, 2003-04-01 at 09:53, Corrado Ventura wrote:
> Hi all,
>
> Anybody know how to configure Asterisk (sip.conf and extension.conf) to
> work with an Adtech Tipcome SIP Phone s-150?
>
> Using a conf like this:
>
> exten => 100,1,Dial,SIP/[EMAIL PROTECTED] (in extension.conf)
>
> and
>
> [a
For those of you running OSX, a new h323 client was released. Haven't
set up h323 yet, so I can't vouch for it.
http://xmeeting.sourceforge.net/
--Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asteris
On Tue, 2003-04-01 at 09:31, Grzegorz Nosek wrote:
> hello all
>
> i'm interested in setting up a small pbx using asterisk and the primary goal
> is keeping the cost down. the general layout of the net is as folows:
>
> * 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be
Hi all,
Anybody know how to configure Asterisk (sip.conf and extension.conf) to
work with an Adtech Tipcome SIP Phone s-150?
Using a conf like this:
exten => 100,1,Dial,SIP/[EMAIL PROTECTED] (in extension.conf)
and
[adtech]
type=friend
username=123
secret=123
host=dynamic
defaultip=192.168.0.
hello all
i'm interested in setting up a small pbx using asterisk and the primary goal
is keeping the cost down. the general layout of the net is as folows:
* 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be
put aside for a traditional phone/fax with no fancy stuff]
* a
On Tue, 2003-04-01 at 03:28, WipeOut . wrote:
> Hi,
>
> I am starting to look at the costs involved in putting a PBX system
> together using *..
>
> We will be using ISDN lines in the UK, which when installed provide a
> box on the wall with 2 analog ports and 2 digital ports, so these seem
> to
On Tue, 2003-04-01 at 08:14, Ahmed Boreau wrote:
> hi,
>
> I'm trying to install asterisk server on mandrake 8.0 and I got a
> dependencies problem with libreadline.so.3
>
> I downloaded readline-4 and readline-2.2.1 package and it still not wrking.
>
> May be some one who got this poblem could
Hey Wade,
I have two and one seems a little more prone to erratic behavior. Now you've
got me wondering if it is something like improper voltage. Maybe the
transformer is slightly undersized and operates just hot enough so the
insulation gradually cooks of the windings - or at least with a little
Based on James suggestion to use the DB functions I made the following and
thought it might be nice to share:
;
exten => s,1,DBget($Night=GlobalSettings/Night) ; if not night jump to +101
exten => s,2,Goto(closed,s,1) ;Night has been set, we're closed
exten => s,102,Goto(open,s,1) ;Night has not be
hi,
I'm trying to install asterisk server on mandrake 8.0 and I got a
dependencies problem with libreadline.so.3
I downloaded readline-4 and readline-2.2.1 package and it still not wrking.
May be some one who got this poblem could help.
thx
___
Aste
I can work with digital E&M - winkstart, immediate, loopstart, groundstart and
ISDN with Qsig. Also with R2, but here in Brasil I prefer the first.
regards
Eduardo
On Mon, 31 Mar 2003 15:17:45 -0600 (CST)
Martin Pycko <[EMAIL PROTECTED]> wrote:
> What signalling are you going to use ?
http://mcleod.pbx.nq.net/msn/
might help to search the mailing list archive through google also
in google type "site:marko.net msn setup"
enjoy
- wasim
On Tue, 1 Apr 2003, it wrote:
> Hi, I would like to ask a silly question. How could I setup MSN messager
> with * ? I really don't
Hi, I would like to ask a silly question. How could
I setup MSN messager with * ? I really don't know how to setup it.
Thanks.
john
After a long working evening yesterday, now my * box place and receive calls
with H323,SIP and ISDN line.
Calling from the office to an outside line, happens:
- If I call a mobile number and the called answers, all goes ok
- If I call a number at home/office, and it's answered , all goes ok
- If
get a t100p and a second hand channel bank from ebay.
the cb will be around 500 US$ which is ok and in my case was a 12fxs/12fxo one
On Tuesday 01 Apr 2003 10:28, WipeOut . shaped the electrons to say:
> Hi,
>
> I am starting to look at the costs involved in putting a PBX system
> together using *.
Hi,
I am starting to look at the costs involved in putting a PBX system together using *..
We will be using ISDN lines in the UK, which when installed provide a box on the wall
with 2 analog ports and 2 digital ports, so these seem to be the options available to
me..
1. I could use an ISDN boa
Hi Vlasis,
CE is no certification, it is just a decleration of conformity
from the manufacturer. It has nothing to do with getting an
ITU / ETSI (whatever...) approval for communication equipment.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Ber
Hello,
I'd like to ask if there are any news about CE certification of the E1
boards. I know that the T1 boards are FCC certified but I'd also like to
know what is the status for CE certification.
Thanks for any input,
Vlasis Hatzistavrou.
___
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