Hi,
> I have problem with Quadbri and bristuffed Asterisk - I guess this is only
> configuration trick. I'd like Asterisk to respond only to 1 number on BRI
> interface and do nothing on other. Right now, even if I leave out that
> number in incoming context, Asterisk takes out and rejects call
Hi,
> gcc -c perlxsi.c -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -
> D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -
> I/usr/lib/perl5/5.8.0/i386-linux-thread-multi/CORE -o perlxsi.o
> gcc: perlxsi.c: No such file or directory
> gcc: no input files
> make: *** [perlxsi.o]
Hi,
Il giorno sab, 16-04-2005 alle 13:30 +0200, Robson Ribeiro ha scritto:
> Frtiz is a nightmare although it is cheap and I have seen it working.
> I have been trying to install it for some days without success but one
> thing is for sure: you have to use the right Kernel (they are
> available fo
Hi,
Il giorno ven, 15-04-2005 alle 18:33 -0400, Ian Pattison ha scritto:
> If I understand your question correctly ztcgf is not a module, it's merely a
> rudimentary diagnostic utility. Run "ztcfg -vv" to get info on your zaptel
> hardware.
Not exactly. ztcfg is the tool that applies the zaptel
Hi
Il giorno ven, 08-04-2005 alle 10:24 +1200, Matt Riddell ha scritto:
> Matteo Brancaleoni wrote:
> > I hate to say that, but the problem is that Digium doesn't do this.
>
> Ahh I beg to differ.
>
> I resell both Digium and Sangoma gear and provide full installation
> support for both.
after
Hi,
> The problem is that when opening the zap channel, originate thinks
> that the call has been answered and send the call to the beginning of
> the context out. And what I really want is to make this but when the
> destiny person answered and not when the zap channel opens.
>
as already in the
Hi,
> Also, the issue i have with incoming calls is odd. I seem to get a
> timeout when dialing my SPA2000. Atleast that is the message. my
> incomeing context is
>
> [incoming]
> exten => s,1,Wait(10)
> exten => s,10,Dial(SIP/3518,20,tr)
/me wonders why s,10
you should use next priority afte
Hi,
> I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
> PBX and ISDN line - if power of Asterisks fails - will those card connect
> PBX directly to ISDN line ?
No, you need a isdn failover switch
> If not are there any other simple switching
> devices, that would do th
Hi,
Il giorno ven, 18-02-2005 alle 12:44 -0600, Eric Wieling ha scritto:
> The Digium Tx00P and TE*xxxP support E&M Wink
E&M is analogue, not digital...
digium cards support it over digital, like they supports fxs/fxo
to a channel bank . same from E&M
The interface described here is analogue, afa
hi,
Il giorno gio, 17-02-2005 alle 22:19 +0100, Olle E. Johansson ha
scritto:
> If you're anonymous, we propably can't match to a user/peer and set
> the
> accountcode from the configuration... Or?
mmmh... but if the user authenticate itself, we can have an accountcode.
I mean anonymous != auth
Hi,
Il giorno sab, 05-02-2005 alle 11:11 +0100, Marcello Lupo ha scritto:
> Hi to all,
> can you suggest to me the best way to avoid problems in the CDRs for
> anonymous
> sip calls?
> I have some peoples that set Send Anonymous : Yes in their Grandstream phones
> and i don't receive the userna
hi,
Il giorno mar, 01-02-2005 alle 13:30 -0600, James Sizemore ha scritto:
> extensions.conf:
> [trunk]
> exten => _X.,1,Dial(${TRUNK}/${EXTEN})
> exten => h,1,Hangup
try
extensions.conf:
[trunk]
exten => _X.,1,Dial(${TRUNK}/${EXTEN})
exten => _X.,2,Hangup
Matteo.
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Hi,
Il giorno dom, 23-01-2005 alle 10:33 +0100, Remco Barende ha scritto:
> What would be the best / easiest VPN software solution. I would like to
> install vpn software on the * server for roadwarriors to connect to with
> laptops running windows. Ideally the vpn solution will not require any
Hi,
Il giorno mer, 19-01-2005 alle 18:45 +, Edin Kozo ha scritto:
> I have a ISDN BRI card with hisax module (w6692) and there is a lot of echo
> when I make calls to outside. Between the sip softphones the echo doesn't
> exist, but when I call to outside through the ISDN the echo exist.
yup
see ${DIALSTATUS} built in var.
can be chanunavail, or busy or what asterisk sets it to.
use it do do your switching.
Matteo.
Il giorno mer, 19-01-2005 alle 00:25 +0100, Rob Scott ha scritto:
> Asterisk seems to regard an unregistered phone to be busy.
> Is that correct? Is not an unregistered p
Hi
Il giorno ven, 21-01-2005 alle 08:54 -0600, Justin Carlson ha scritto:
> is the hint
>
> 99,hint,ZAP/1
that works only for sip channels.
if you want hint working also for zap, you should
check very latest bristuff at junghanns.net website.
Afaik he has added (among support for bri cards)
exte
Hi,
Il giorno mer, 12-01-2005 alle 00:38 +0200, Shoval Tomer ha scritto:
> Only if you don't have Digium hardware installed.
yes
> And only for MeetMe, I think.
>
> Correct me if I'm wrong on this, though...
really, it works for zaptel timing, that's needed only
by meetme and iax2 trunking. But
hi
> i have HFC supported ( Planet TA ) card installed on Redhat 9 and i have
> installed bristuff
> and i can load zaptel and load zaphfc module in TE mode . and unable to load
> ztdummy module properly
you don't need ztdummy if you have a zaptel card installed
> here is my zaptel.conf
>
> loa
hi,
> I receive a call at the extension. Press the hold button. Music on hold
> starts. When I place the handset back on the cradle, the call gets hung
> up/disconnected. The Phone is A GrandStream Budge Tone 100.
this seems a phone problem.
2 solutions:
* don't put the handset back on the cradle
Hi,
Il giorno gio, 16-12-2004 alle 21:47 +, Jean-Michel Hiver ha
scritto:
> I was wondering if there was any device I could use to connect * to GSM
> networks. I don't need much capacity, maybe 2-4 GSM channels. As usual,
> cheap is better :-)
sure, mainly you can use gsm boxes with pstn t
Hi,
Il giorno gio, 16-12-2004 alle 21:59 +0400, Muhammad Talha ha scritto:
>
> Dear all
>
> i am using Fedora Core 2 . i have Planet BRI TA with HFC chipset ( hisax )
> i can easyly connect to internet using BRI but this card is still not
> recognized by asterisk i am using i4l driver .
don't
Hi,
Il giorno dom, 12-12-2004 alle 14:38 +0200, Warren Burstein ha scritto:
> When I first saw the priority numbers in extensions.conf, I thought BASIC,
> if a number is missing, * will fall thru to the next number. I learned that
> this is not so, if you have nothing between 1 and 3, you don't e
Hi,
Il giorno dom, 12-12-2004 alle 00:36 -0800, Charles S. Antrim ha
scritto:
> I have success installing and compiling, but if I reboot I have to modprobe
> again to get he
> drivers loaded for the module I am using. I am using rhes31 and a tdm card
> with one fxo and
> one fxs.
perhaps you
hi,
Il giorno mer, 01-12-2004 alle 15:37 -0300, Listas ha scritto:
> Hi I would like to know which is the last stable version of asterisk and how
> to get it from the CVS, I mean rather than doing
>
> cvs checkout -r -v1-0_stable asterisk
go on asterisk ftp site and look for 1.0.2 tgzs,
or co vi
Hi
Il giorno mer, 24-11-2004 alle 19:48 +0100, Ming-Wei Shih ha scritto:
> Hong Kim wrote:
>
> >I'm running * on Redhat9 with E100P and ISDN PRI.
> >When I executed asterisk, I could see about 25
> >asterisk processes.
> >Did someone experienced this?
> >
> >Regards,
> >Hong
> I only see one :)
Hi,
Il giorno mar, 23-11-2004 alle 16:38 -0500, Jay Brussels ha scritto:
> I started out with the development branch then switched to the stable (as the
> entire company now runs on Asterisk).
> The stable branch (including 1.02) does not have the queue annoucements.
I'm sorry to contradict yo
Hi
Il giorno dom, 21-11-2004 alle 20:49 +, Mike Dent ha scritto:
> Ok, so I realised I was running a CVS version of * which might have been
> giving
> me the SIP problems. So I decided to get down 1.0.2. I followed the usual
> instructions, compiled and installed it. (FC2)
> [chan_zap.so]
Hi,
> Y si te molesta no hubieras respondido OK ?
because this isn't a nufone support ML.
The next time, post your configs, not
your complains about nufone. Without that,
no one has divinatory powers and can help you.
> y mas BLA BLA BLA eres TU ... porque quizas
> cuando TU no habias nacido aun,
Hi,
> Nufone provide me some config examples ... I can dialout
> but I can't register my * Box, eg. whe I do "iax show registry"
> I got only a "Request Sent" and later I have a "Timeout"
First of all, I'm (and many are) sick to see blah blah
blah doesn't work with blah blah blah.
This is * M
Hi,
> I have a PRI card. How do I block a caller id sent out to PSTN from a
> SIP client? I add a remote-party-id field "privacy=full" but still get
> caller id on a PSTN phone.
I think that doing SetCIDNum() (with no args)
before dial will do the trick.
Matteo
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de in dundi.conf.
comment it out on dundi.conf and see www.dundi.com
to learn what is dundi :)
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alize the call is answered. After 15 seconds it
> proceeds to voicemail interrupting the call. Can anyone help?
eh, perhaps with some details about your zap...
ie what card?
zaptel.conf?
zapata.conf?
matteo, still without divinatory powers
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*/libpri/zaptel euroisdn bri (from klaus)
and us bri could be a great idea. is of course a bigger plus
for * itself
matteo
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meone that can
do for you.
Is not a binary file, don't you agree???
matteo.
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To
te... my english is getting worse :(
sorry for it.
matteo
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that's not a problem :)
Matteo.
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st a connection.
Using * + mysql 4.1 since when mysql 4.1 was in beta.
also used asterisk + mysql cluster for a while, only
in the lab, but never lost connection also
in that case.
Matteo.
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will want to add checks for other services,
I currently use it to monitor also apache, mysql,
crond, sendmail, etc etc ect
matteo.
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http:/
a process (normally spawned by init) that
checks other processes.
I suggest you monit (http://www.tildeslash.com/monit/),
it can monitor processes in various ways and other
vital system informations. I'm pretty happy
with it (using it in all my * installations)
Matteo
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be 100% sure.
matteo.
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Hi,
> Are there any solutions to avoid cdr manipulations
> by users, who prepare special caller id strings?
set the callerid from asterisk. don't let others
to set it.
Matteo.
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_
how many fd has your asterisk open
with something like "lsof | grep pipe | grep asterisk"
Matteo.
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card, that does E1 or T1 (like the quads)
and has a new board / design (perhaps
to be able to certify it like the quads)
Matteo.
P.S. I think that if you order now, you'll get
the new cards.
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Hi
Il lun, 2004-10-11 alle 18:47, shabanip ha scritto:
> btw, can i read them from agi? how?
use "get variable" agi command, like "get variable foo"
Matteo.
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r to not send any noise and/or echo.
I'll try to keep the link to nufone conf room
up until astricon end.
Matteo.
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7;t remeber where is,
now.
matteo.
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* [common.o] Error 1
> make[1]: Leaving directory `/home/asterisk/asterisk-addons/format_mp3'
> make: *** [format_mp3/format_mp3.so] Error 2
>
>
>
>
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] O
ort
OR
* try to resolve the issue and inform the ml
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call spooling file to connect
to the meet me room and play any file
with playback,background,mp3player,blah
matteo.
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mmunity is pretty good about keeping packages updated.
ah! ah! ah!
really... oh oh, so why debian is eons later in releasing
new packages...
perhaps you're speaking of -unstable debian... that's
wy too unstable.
btw, I'm only joking... nothing serious here :)
Matteo.
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terisk in part
of their network
> Are there some problems with asterisk ???
if used wisely, ie "you know what you're doing" , no
Matteo
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t doesn't have the divination plugin,
so please report you errors.
a mail like that is only annoying,
thanks.
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sdn&co is in libpri, r2 should be in libr2, but
is far from being complete.
Matteo
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To
xs card.
on the bugtracker there's a patch that allows to raise
loopcurrent on the proslic, feel free to test it.
has resolved many issues with third party devices.
Matteo.
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sorry, I misread your post.
check from asterisk console:
show manager commands
if the function getvar is registered.
here with rc1 works without probs.
Matteo.
Il mer, 2004-07-21 alle 19:13, Brancaleoni Matteo ha scritto:
> dialplan apps are not manager apps
>
> matteo.
>
>
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isk in it's full glory with agi support to handle 1 fxo, 1
> fxs, and sip off to a provider such as voicepulse.
>
> Eric
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I uploaded every RC1 stuff to
http://asterisk.espia-net.net/asteriskRC1/
just to help the digium slow link.
Matteo Brancaleoni
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'll see all codecs supported, along with translation tims
> >
> I believe Asterisk has the same codec list in all of its versions.
> Well, at least for the versions I've seen.
>
> I'm sure someone will rush to correct me if I'm wrong.
as far as I know, the st
hi...
here in Italy is almost impossible to set an
invalid cid, if is out of your allowed space.
ie. if you have X numbers on your PRI,
you can only set one of these. nothing more.
on bri you simply cannot do nothing.
just my 2 cents.
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Esp
is not part of current play.
but is easy to add info to iax to carry what you need.
> what am i missing here?
experience.
btw, SIP is certainly needed 'cause of the clients...
much more available than iax ones.
but for server to server pov, iax is sure a better choice.
Matteo.
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Hi
Il lun, 2004-07-05 alle 20:12, Brian K. West ha scritto:
> *8# works on sip that uses the # as the send key.
sure, but since he gets
-- Sent into invalid extension '*8#' in context 'from-sip-post'...
means that he's sending *8# ...
matteo
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via *8?
>
> Adolfo
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e=friend
> host=dynamic
> username=ciscok
> canreinvite=no
> callgroup=2
> pickupgroup=2
> mailbox=100
> qualify=1000
> dtmfmode=rfc2833
> trunk=yes
>
> [ciscok2]
> type=friend
> host=dynamic
> username=ciscok2
> canreinvite=no
> callgroup=2
pe=friend
> username=damencho
> host=dynamic
> nat=yes
> canreinvite=no
>
>
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>h
hi
> Can someone please confirm that their E100P says "T100P" on the artwork?
yes, the board is the very same. the only difference
is in the framer chip.
btw, just plug it, build zaptel & load wct1xxp module
and you'll see that is an E1 card. (in dmesg)
Matteo.
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.
'T' -- allow the calling user to transfer the call by hitting #.
Doesn't that work?
Matteo
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> And I did this after moving the current zaptel, asterisk, and libpri to
> archival.
>
> Where do I get this file?
> Or what am I doing wrong...
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date *should* be generated by the fxs device.
When I receive cid on my analog phones via digium fxs cards,
the time is kept up in sync with the server.
Matteo.
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.de/english/downloads/isdn/saphir_5_primary_pci/index.asp
unfortunately they provide only precompiled kernel modules,
and only for certain kernel versions... so you're stick
to what they use.
Matteo.
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e0", "ZAP/g1/h") in new stack
> -- Called g1/h
> -- Channel 1, span 1 got hangup
> Jun 11 23:39:43 WARNING[491541]: app_dial.c:349 wait_for_answer: Unable to forward
> voice
> Jun 11 23:39:43 WARNING[491541]: app_dial.c:349 w
he same way.
so the easy way it to rsync /var/spool/asterisk/voicemail
to the backup server, so if the primary goes away,
you'll have your voicemails as before.
of course when the primary is back a sync back must be
done...
matteo.
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Hi
Il ven, 2004-06-04 alle 18:54, Christopher Wall ha scritto:
> Where do I identify the listen port on my asterisk box?
depends on what service?
sip uses 5060 in sip.conf
manager 5038 in manager.conf
iax2 4569 (hardcoded, changeable from chan_iax2.c)
blah blah blah :)
Matteo
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the network cards modules
will be what you'll need to do...
Matteo
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Hi
> I was planning to use the output of asterisk -rx "show queues" in a
> script when I noticed that sometimes asterisk only outputs the first
> line of the response. e.g:
why don't you use the manager interface?
it's much better...
Matteo
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as Tony suggests, rebuild the kernel first, install it and then
build zaptel
Matteo.
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led... like php,perl,bash scripting,ruby,
whatever
AGI speaks with the app with stdout/stdin/stderr ...
so anything that supports this IO can be used :)
Matteo
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open ports 1 to 2 UDP as in default rtp.conf
isn't a problem, since there's not any port open...
(unless you run any udp service on that interval :) )
and a portscan will detect these port as closed.
only during a call, * and the phone will handshake an RTP
port and use that
setting into the extension seems to me the same as setting
into iax.conf (or sip.conf), or not?
otherwise... use very strange passwords along with superstrange
usernames I bet someone to get a login data like
username : 2h729872pcnt
with pw : inr2.f2f2232DDFW3r
or not :) ?
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all
please double check that (as I'll do...)
Matteo.
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right sections of the dialplan.
> >
> >exten => ,1,goto(Sales-in,s,1)
> >exten => ,1,goto(Tech-in,s,1)
> >exten => ,1,goto(vmail,s,1)
> >exten => ,1,goto(extensions,110,1)
> >exten => ,1,goto(extensions,111,1)
> >
> >G
rs mailing
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Brancaleoni Matteo <[EMAIL PROTECTED]>
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eems ok. very simple,
but does the job.
Matteo.
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odems here)
with standard modem, you can build an echo generator
and latency driver.
modem driver under * isn't fully supported, since
there're better ways to do voice.
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> > Lug-nuts mailing list
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> allow=ulaw; Allow all codecs
>
> Lisa
>
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ur support!
>
> /Olle
>
>
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- I think those should be Digium's flagship products,
> not a rebranded craptastic WinModem.
Hope so.
surely works better than the intel one, and I don't see any reason
in loosing (your, of course) time into making it work under zaptel.
Isn't it a craptastic WinModem also? even
odule parameters,
> including invalid IO or IRQ parameters.
> You may find more information in syslog or the output from dmesg
>
> is there anything more I can do?
> tia
> mazek
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Brancaleoni Matteo <[EMAIL PROTECTED]>
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alle 16:46, Ryan Thrash ha scritto:
> FYI, with 1.0.4.55 and NAT set to off (but with the * config set as
> nat=yes), I'm able to bypass stun servers completely with a GS phone as
> well.
>
> HTH,
> Ryan
>
>
> On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote:
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as snom200.bin
then reboot the phone, and as soon as it powers up
(don't let the phone boot at all), press a key.
it will prompt an ip addr,netmask,gw and tftp server addr.
fill the values and go on.
the phone will load the firmware, and everything
will be set @ default values.
Matteo.
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Hi
>
> >I made a custom fedora mini distro, something like
> >350 megs, including apache,php,mysql & webmin
> >
> >of course installable from a cd in 20 minutes, more or less :)
> >
> >at the end you have a fully working asterisk installations,
> >along with some basic tools like webmin and
> >a f
I made a custom fedora mini distro, something like
350 megs, including apache,php,mysql & webmin
of course installable from a cd in 20 minutes, more or less :)
at the end you have a fully working asterisk installations,
along with some basic tools like webmin and
a full webserver
Matteo.
Il
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