Hi Garet,
ok but since the messages contain my own public IP with this method I'm
banning my public IP not the real attacker IP. Am I wrong?
Giorgio
On 10/01/2013 05:26 PM, Gareth Blades wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal
.
in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject
1st invite.
On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.
Is there a way to fix this problem?
Thank you.
Giorgio
a predefined number of (failed) authentication attempts.
Regards,
Ricardo
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com
wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
Hi Shitian,
the line works but the ERROR is annoying since it appears very
frequently. I think I'll have to patch it in order to lower its
priority, maybe a NOTICE.
G
On 02/22/2013 03:06 PM, Shitian Long wrote:
Did you get it to work may I ask ?
On Feb 20, 2013, at 3:49 PM,
Hi all,
has anybody ever encountered this ERROR before? It happens frequently on
my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1
and a quadBRI card.
ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured
Component
I tried to google but without
Hi all,
the problem has been solved setting pridialplan=unknown.
Asterisk restart is mandatoryreloading chan_dahdi.conf module is not
enough!
The leading zero still remains a ghost since it cannot be seen in any
log. B! Creepy! :)
Hope this can help some other soul in pain...
*Subject:* Re: [asterisk-users] leading ghost 0
Not only, you have to restart dahdi/zaptel as well.
Leandro
2012/11/20 Frederic Van Espen frederic...@gmail.com
mailto:frederic...@gmail.com
On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
I'm sure nobody has added something... tried
] added by Asterisk/DAHDI??
I've used this page as reference about frame fields:
http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm
Thank you.
Giorgio Incantalupo
On 11/20/2012 05:23 PM, Alex Kauffmann wrote:
On 11/20/2012 8:03 AM, gincantalupo wrote:
Hi Leandro,
I'm sure nobody has added
Hi all,
I have problems dialling out because my new telco (the previous gave no
problems) tells me my PBX adds a leading 0 and that's why I cannot dial
out (but I can receive calls).
I make a small extensions.conf as a test:
exten = 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out
wrote:
2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com
Hi all,
I have problems dialling out because my new telco (the previous
gave no problems) tells me my PBX adds a leading 0 and that's why
I cannot dial out (but I can receive calls
are using?
Leandro
2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com
Hi Leandro,
thanks for your answer.
I already have tried those parameters but without any positive result.
The telco technician has tried the line with its machine
Hi Matthew,
you are right...it seems that extensions.conf behaviour has been changed
from asterisk 1.4.
Thank you.
Giorgio Incantalupo
On 10/03/2012 05:40 PM, Matthew Jordan wrote:
- Original Message -
From: gincantalupogincantal...@fgasoftware.com
To: Asterisk Users Mailing
Hi guys,
I've upgraded my pbx from asterisk 1.4 to 1.8 but parking does not work
anymore. Tried asterisk-1.8.11.0 and then, after reading about a (fixed)
problem in CHANGELOG tried asterisk-1.8.16.0, without success.
My features.conf is:
[general]
parkext = 700 ; What ext. to dial to park
Hi Matthew,
this is the result of dialplan show:
's' =1. NoOp()
[app_dial]
[ Context 'parkedcalls' created by 'features' ]
'700' = 1. Park()
[features]
[ Context 'macro-hash-automon' created by
.mclink.it
insecure = very
language = it
fromuser = 123456789
username = 123456789
secret = passwd
On 07/02/2012 12:32 AM, Shitian Long wrote:
if you check out your sip.conf.
On Jun 29, 2012, at 5:54 PM, gincantalupo wrote:
Hi all,
after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register
Hi all,
after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my
VoIP provider because it says I'm trying to connect to port 55150
(that's what the call center guy told me)...but I'm not. In my sip I've
set port=5060, not 55150.
The strange thing is that the rport inside SIP
T.38
gateway. The result was the fax was retried for the defined number
of attempts.
Cheers,
Larry.
On 16/05/2012 6:28 PM, gincantalupo wrote:
Hi all,
I'm trying to lower my iaxmodem speed but still I haven't found any
solution...I tried to add
Class1RMQueryCmd: !24,48,72
Hi guys, thanks for answers.
That could seem counter-intuitive but it is not. Not to mention the fact
that information technology is not science, the solution to broken faxes
is to lower down speed. This works even with normal telco lines even if
you DO NOT have a pbx (telco technicians even
Hi all,
I'm trying to lower my iaxmodem speed but still I haven't found any
solution...I tried to add
Class1RMQueryCmd: !24,48,72
to config.IAXtty but does not work...Hylafax says it it running at 9600
(sometimes at 14400) baud..
Any ideas?
Thank you.
Giorgio
--
number of
attempts.
Cheers,
Larry.
On 16/05/2012 6:28 PM, gincantalupo wrote:
Hi all,
I'm trying to lower my iaxmodem speed but still I haven't found any
solution...I tried to add
Class1RMQueryCmd: !24,48,72
to config.IAXtty but does not work...Hylafax says it it running at 9600
(sometimes
Hi Larry,
I forgot to mention I tried to set ModemRate at 4800 as well but without
success.
Giorgio
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
hylafax/config/iaxmodem I added:
Class1RMQueryCmd: !24,48# V.17 fast-train recv doesn't
work well
Class1TMQueryCmd: !24,48# V.17 fast-train recv doesn't
work well
Giorgio
On 05/16/2012 05:33 PM, Patrick Lists wrote:
On 16-05-12 17:10, gincantalupo wrote:
Hi Larry
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi Alex,
replace with anything which could make Asterisk connect to Skype
network, make and receive calls, etc...the usual stuff.
Giorgio
On 12/01/2011 02:40 PM, Alex Balashov wrote:
On 12/01/2011 08:30 AM, gincantalupo wrote:
any idea about how to replace Skype For Asterisk?
Replace
.
On Fri, Oct 14, 2011 at 6:21 AM,
gincantalupo gincantal...@fgasoftware.com
wrote:
Hi all,
I'm stuck on a tricky problem.
I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom
Phones. When I call an IVR I get the damned one
Hi,
found where the problem is.I tried with a Grandstream phone and it
works!!!
The problem is my (crappy) Snom phone.
I'm investigating the probhope to find the cause asap.
Sorry for wasting your time, guys. :)
Giorgio
On 10/14/2011 12:21 PM, gincantalupo wrote:
Hi all,
I'm
Hi all,
I'm stuck on a tricky problem.
I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When
I call an IVR I get the damned one way voice phenomena. It is not
randomic, it happens all the time.
I tried to upgrade the snom firmware to 7.3.30 but nothing changed.
If I call a
Is there a dahdi_cfg in your boot sequence? When I modify dahdi config
files I always launch dahdi_cfg otherwise I get errors like yours.
Giorgio
On 06/14/2011 05:37 PM, Olivier wrote:
After a reboot, I can't reproduce the problem anymore which is quite
frustating.
2011/6/14 Tzafrir Cohen
Hi,
it is a known problem, one of the worst. To avoid it:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially sip
providers urls)
or install a dns-cache on your pbx (maybe the best solution)
Giorgio
On 05/30/2011 03:10 AM, nhadie
AM, gincantalupo wrote:
I could create 2 queues, one for italians and one for strangers
calling but there is no point where you can change the moh except
before executing the queue command but the queue moh changes as
side-effect:
Hmm. When you use SetMusicOnHold, does it change the queue MOH
Hi all,
I want to change my moh without changing my queue music...is it possible?
SetMusicOnHold changes my moh but with the wrong effect to change my
queue music I do not want to change...did anybody solve this problem or
it is a bug?
Thank you
Giorgio Incantalupo
--
, gincantalupo wrote:
I want to change my moh without changing my queue music...is it
possible?
SetMusicOnHold changes my moh but with the wrong effect to change my
queue music I do not want to change...did anybody solve this problem or
it is a bug?
Just define different MOH classes
:
On 05/18/2011 11:12 AM, gincantalupo wrote:
I tried but doesn't work because the queue doesn't give you control
to change the moh. What I want is to change my moh depending on where
the call is from. If it comes from Italy I have to play italian moh,
if not, another moh. Normally I can change my moh
Hi,
I'm trying Yealink phones too but I cannot provide remote assistance to
our customers using a text-based browser like lynx (I know I could use
(t)ftp provisioning system but my boss does not like it).
Any idea or work-around?
Thank you.
Giorgio Incantalupo
On 02/25/2011 06:04 PM,
Hello,
I tried this piece of extensions on my Asterisk 1.8:
exten = 679,1,NoOp(start)
exten = 679,2,AGI(/var/lib/asterisk/bin/test.py)
exten = 679,3,NoOp(--- end ---)
exten = 679,n,Hangup
where test.py executes a queue command.
The strange thing is my CLI never shows the '--- end ---' string.
Hi,
every time I park a call, my Asterisk box restarts.
This happens during calls made with AGI (not with normal Dial command).
Seems to be an Asterisk 1.4(.17) problem since my Asterisk 1.2(.18)
works fine.
Anybody got some hint?
Thank you!
Giorgio
, gincantalupo
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Hi,
sometimes Asterisk drops calls and shows Didn't get a frame from
channel in its log file. Unfortunately Google gives no answers
even if
a lot of people ask for help.
A fast look into the code shows
Hi,
sometimes Asterisk drops calls and shows Didn't get a frame from
channel in its log file. Unfortunately Google gives no answers even if
a lot of people ask for help.
A fast look into the code shows Asterisk entering a loop where voice is
been transferred and every loop Asterisk waits for a
)
WARNING[10383]: chan_sip.c:1968 retrans_pkt: Hanging up call
[EMAIL PROTECTED] - no reply to our critical
packet.
How did you solve this problem?
Thank you.
Giorgio
Ex Vito wrote:
On Thu, May 29, 2008 at 11:00 AM, gincantalupo
[EMAIL PROTECTED] wrote:
Hi Ex Vito,
I took a look at some
Hi Tharanga,
you could tell the receptionist to press *0 (see features.conf for
details) to take the call again before it is sent to the voicemail.
Giorgio
Tharanga wrote:
Hi List,
Iam using asterisk 1.2.14. when someone call to my ip-pbx, receptionist will
transfer that to specific
Hi,
hope not to be OT :)
after more than 3 years of PBX installations we can adfirm Asterisk is
stable enough to be considered a good product but still we encounter a
lot of problems when deploying a new PBX. It seems that the biggest
problems are all networking related: one way voice (also
make tests and let we know the results.
Regards
On Tue, May 13, 2008 at 7:29 AM, gincantalupo
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Hi,
I'm making some tests with Tellfree brazilian provider. I'm using 2
users A and B, one for calling and the other to receive
Hi Ajey,
which kind of BRI are you using?
Giorgio Incantalupo
Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
Regards
Ajey
Hi Faraz,
yes, you can use ztdummy but it cannot completely replace Digium cards.
It depends from your hardwareI had troubles with some kind of
serversso beware.
Giorgio.
Faraz R. Khan wrote:
You can do conferencing without the zap interface. just modprobe
ztdummy. Its good for small
Hi Mark,
you can use Asterisk with a PRI card. Just set channels signaling to
pri_net in zapata.conf.
Mark Welch wrote:
Hello All,
Does anyone know of a software emulator that can be used to simulate
hardware such as an E1? I need to play with AstUnicall in a test
environment and don’t
Hi Kevin,
unfortunately I live in Italy and you is not so easy for us to get
electronic stuff.
Let's wait and see what happens.:)
Giorgio
Kevin P. Fleming wrote:
Giorgio Incantalupo wrote:
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The
Hi Steve,
I've already tried a Sangoma card and it behaves the same as TDM400P.
But the problem arises for example when I have to change a broken card
on an old PBX keeping the modules, that's why I need a clone card like
Openvox (Sangoma modules are different as you know) Moreover I'd like
Hi Olle,
that was a phone misconfigurationa parameter had a wrong value.
The message has disappeared and now the phone seems to work!
Thank you!
Giorgio
Johansson Olle E wrote:
10 jan 2008 kl. 16.48 skrev gincantalupo:
Hi,
I'm using an Asterisk 1.2.18 box with a remote Snom 360. My
Hi,
I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always
rings but sometimes (it happens randomly!) no voice is passing thru (2
ways).
Asterisk CLI shows this warning:
Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on
authentication for INVITE to
Hi,
I have an Asterisk PBX 1.2.18 connected to a remote Snom 360 (firmware
ver: 6.5.10), outside my LAN. If I call the Snom, I sometimes cannot
hear the called party and the called party cannot hear me. When this
happens, I get the following message from Asterisk CLI:
NOTICE[19164]:
Hi Steven,
I do not live in China but I had the same problem.
Try these 2 params inside zapata.conf:
busydetect = yes
hanguponpolarityswitch = yes
It worked for me.
Giorgio Incantalupo
Steven O'Reilly wrote:
Afternoon,
I was hoping someone could point me in the right direction. I
Hi,
why not using the last version?
Giorgio
fateme fatah wrote:
Hi:
I want to use asterisk1.2 but I don't know which version of
asterisk1.2 and zaptel1.2 is best.Please offer me one version of
asterisk and zaptel and libpri.How about asterisk1.2.24 and
zaptel1.2.20.1 and libpri1.2.5?And
Hi Chawki,
it is not uncommon that FXO or FXS modules do not work even if no error
message is shown when wctdm module is loaded.
Have you tried to replace/swap your FXO modules?
Giorgio
chawki hammoud wrote:
Hi:
I have a digium tdm04B. One fxo module always gives a
faulty busy signal when
Hi Tzafrir,
Tzafrir Cohen wrote:
On Thu, Sep 13, 2007 at 05:32:25PM +0200, gincantalupo wrote:
Hi,
I've installed Asterisk with a TDM400P on a Debian Etch distro.
When I reboot the server I get zaptel and wctdm automatically loaded.
I'd like to avoid this behaviour.
Why exactly
with my init script.
Giorgio
Tzafrir Cohen wrote:
On Mon, Sep 17, 2007 at 09:36:27AM +0200, gincantalupo wrote:
Hi Tzafrir,
Tzafrir Cohen wrote:
On Thu, Sep 13, 2007 at 05:32:25PM +0200, gincantalupo wrote:
Hi,
I've installed Asterisk with a TDM400P on a Debian Etch
Hi Tzafrir,
Tzafrir Cohen wrote:
On Mon, Sep 17, 2007 at 10:23:53AM +0200, gincantalupo wrote:
Hi Tzafrir,
I'm currently using genzaptelconf for digium cards but I have Sangoma
cards too and I have to load digium drivers after Sangoma one.
That's why I need to understand why wctdm
Hi,
is there anybody using an analog TOPEX GSM gateway?
My asterisk + TDM400P does not receive the hangup signal from that gateway.
Is there anybody who can give me a hint?
Thank you!
Giorgio
___
Sign up now for AstriCon 2007! September 25-28th.
Hi,
I've installed Asterisk with a TDM400P on a Debian Etch distro.
When I reboot the server I get zaptel and wctdm automatically loaded.
I'd like to avoid this behaviour.
How can I disable this automatic modprobe wctdm during boot?
Thanks
Giorgio
Hi satish,
I get that error too (my Asterisk version is 1.2.x but should be the
same) when that Zap channel is not available and you are trying to use it.
You should get a CHANUNAVAIL from Asterisk channel status.
Giorgio.
satish patel wrote:
Dear all
I have asterisk 1.4.11
Hi,
I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium
TDM2400P.
I got this error inside /var/log/messages:Power alarm on module
8, resetting!
I rebooted the PBX and this time I got:Power alarm on module
7, resetting!
Please, does anybody know what it
Hi all,
I solved the problem changing the module.
Giorgio
gincantalupo wrote:
Hi,
I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium
TDM2400P.
I got this error inside /var/log/messages:Power alarm on module
8, resetting!
I rebooted the PBX and this time I got
Hi,
I'm trying to connect an HDL F10 device for a friend living in Brazil to
the TDM2400 on his Asterisk server.
That device should behave like a normal doorbell and it is if connected
to an analog PBX.
I connected to the TDM2400 and everything works fine except for one
thing: when the called
wrote:
gincantalupo wrote:
Hi,
I'm trying to connect an HDL F10 device for a friend living in Brazil to
the TDM2400 on his Asterisk server.
That device should behave like a normal doorbell and it is if connected
to an analog PBX.
I connected to the TDM2400 and everything works fine
Hi Mauro,
changing from pmtp to ptp port type increased the time to call outside.
Log files show the card waiting for something from the telco line.
What kind of provider and ISDN port type have you got?
Giorgio
Mauro Zanin wrote:
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron
Hi,
I've compiled rxfax and txfax on my Asterisk box.
I've made a small extensions.conf for test so that when I call a number
with my Idefisk softphone I activate txfax command.
My goal is to try to send a fax using an analog line first and then an
ISDN line but I do not know how to specify the
Hi,
I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on
it...only a TAE connector.
I'd like to create an adapter so I need to know which TAE pins to
connect to RJ 11 pins.
Is there anybody who knows where I can find a schema of that adapter?
Single connector pinout may help
Hi,
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback:
Failed to write frame
-- Playing 'beep' (language 'it')
Jul 5 10:42:32 WARNING[23960]:
Hi Noah,
1 - my asterisk version is 1.2.18
2 - my SIP devices are SNOM phones
3 - no SIP provider is involved...they are connected to my
Asterisk...this is the strangest thing.
This happens sometimesI think it could be a network overload...can
it be?
TIA
Giorgio
Noah Miller wrote:
Hi
as unreachable by asterisk box, and router is
reset on remote location, phone reregisters.
Any help is appreciated.
Thnx
Mihaela MJ.
On 7/3/07, *gincantalupo* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi Olivier,
I forgot to mention it is a C450IP.
But if you have some
Hi Olivier,
I forgot to mention it is a C450IP.
But if you have some hint on S maybe it can help me. Perhaps it is some
configuration...I tried with qulify=no as I read on a web page without
success.
Thank you.
Giorgio Incantalupo
Olivier wrote:
Is it a S 450IP ou C 450IP ?
Hi,
somtimes my Gigaset 450IP loses its registration.
Is there anybody who knows why and how to solve it?
TIA
Giorgio Incantalupo
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or
Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box. I've noticed a very
strange fact which causes a bad prob. When I get an inbound call, I make
4 phones ring at the same time, one is a Snom while others are Gigaset
C450IP with _latest firmware_.
When I get a call and answer with the Gigaset, a
Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box.
I have two phones a Gigaset C450IP and a Snom 360. Suppose someone is
calling the Gigaset phone and a second call comes and is redirected to
the voicemail: if the new caller hangs up during voicemail announcement,
Asterisk drops the first
Hi Malcom,
my advice is to run asterisk as non-privileged user. And do not user
safe_asterisk script if u can...it cannot check if it running and you
can have many safe_asterisk running in memory...moreover this fills your
CLI with a lot of annoying remote unix connection messages.
Giorgio
Hi,
I cannot get attended working on my Asterisk 1.2.9.1 during an inbound
call via an ISDN card to a Snom SIP phone.
The called party is not able to transfer even if :
1 - atxfer is enabled (set to *7) in in features.conf
2 - the dial option is set to value 't'
3 - I see * and then 7 on
Hi Chris,
we tried TDM800 and TDM2400 without problems even if it is a pity not
to have leds on the card (as TDM400).
BTW I think it is better to have only one card on PBXs, when possible of
course!!
Giorgio
Chris Earle wrote:
Hi all,
Years ago, I was pretty sure attempting to use two
Hi,
I tried to patch asterisk 1.2.1 on a Debian Sarge distro with the patch
made by tzafrir but I still cannot set writing permission to directories.
I tried to put umask 007 inside .bash_profile but it doesn't work.
Is there anyone who can help me?
TIA
Giorgio Incantalupo
Hi,
we are trying a beronet ISDN card with asterisk 1.2 on debian sarge distro.
Everything seems fine except for outbound calls: it seems we cannot send
outbound digits so we cannot use phone digits to use ivr menus.
I followed beronet dinstallation document.
Is there some parameter missing to
Hi all,
problem solved!
The parameter /s at the end of Dial string command was necessary.
Giorgio Incantalupo
gincantalupo wrote:
Hi,
we are trying a beronet ISDN card with asterisk 1.2 on debian sarge
distro.
Everything seems fine except for outbound calls: it seems we cannot
send
Hi Marcus,
haven't you got an Unable to initialize mISDN error during asterisk
startup?
I have a problem with chan_misdnI'm trying to understand where is
the prob...I haven't recompiled my kernel with mISDN support because
Digium claims it is inside Asterisk 1.2, maybe it's my
Hi,
I have the same problem, same error but loading modules changes nothing.
I'm using debian sarge and Asterisk 1.2: after compiling asterisk I
launched install-misdn from beronet site.
When I started Asterisk the same error arose:
Nov 30 15:43:06 ERROR[4914]: chan_misdn.c:3455 load_module:
Hi,
I'm setting up an Asterisk 1.2 PBX based on a Debian Sarge distro with a
quadBRI beroNet card.
I've followed beroNet instructions so I compiled Zaptel, Libpri and
Asterisk and then launched install-mISDN script downloaded from beronet
site (install-mISDN.tar.gz).
I try to start Asterisk
Hi,
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm
using a K8N-E deluxe asus motherboard which gives me some problems (but
I'm not sure is the motherboard causing the problem):
- if I plug a TDM400 REV J, Debian cannot recognize it
- if I plug a TDM400 REV E/F,
you tried latest drivers?
could be simply a pci-id problem.
matteo.
Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto:
Hi,
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm
using a K8N-E deluxe asus motherboard which gives me some problems (but
I'm
should be a better way to solve the problem but
I don't know where to find that damned sequence.
Giorgio Incantalupo
This
Tzafrir Cohen wrote:
On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote:
Hi,
I had some problems to with a quadBRI with a 2.6 kernel debian distro.
Have you
Hi,
I had some problems to with a quadBRI with a 2.6 kernel debian distro.
Have you tried to insmod the zaptel.ko module instead of modprobing?
It worked for me, hope it will work for you too.
Giorgio Incantalupo
Remco Barende wrote:
Hi list!
On a newly installed RHEL 4 box I'm trying to
Hi Angus,
I have the same problem but on a Debian distro I do not know very well...
When I boot the machine only wcfxs and zaptel modules are loadedhow
can I load qozap before wcfxs?
TIA
Giorgio
Angus Comber wrote:
Hello
I am sure this is a very basic Linux question.
But every time
Hi,
I'm attempting to run * as a non-root user (asterisk), I can run * as
my new user with /usr/sbin/asterisk -c without problem.
However, I'm unable to run * using safe_asterisk with my user, the error
shown is:
Asterisk ended with exit status 127
safe_asterisk is trying to
-info.org/tiki-index.php?page=Asterisk%20non-root
If it still does not work, append the last part of your
/var/log/asterisk/messages
Christian
gincantalupo schrieb:
Hi,
I'm attempting to run * as a non-root user (asterisk), I can run *
as my new user with /usr/sbin/asterisk -c without
Hi zoltan,
I have got the same problem...same error. Seems like the makefile is
searching for a modules rule but I looked into Makefile and there is
not a 'modules' rule...
Have you found a solution?
TIA
Giorgio
Zoltan Szecsei wrote:
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14,
Hi,
[EMAIL PROTECTED] wrote:
Is there a way to
1) disable asterisk from writing in the full log ? (
/var/log/asterisk/full )
Take a look at /etc/asterisk/logger.conf
or
2) implement a log rotation or similar of the full log ?
I see the full log grows a lot (about 100 MB per
Hi,
I could be wrong but...have you taken a look at features.conf??
Usually settings like the one you need are stored there.
Giorgio
hak atil wrote:
How can I override *67 to *8? Is there an easy way to do this?
Thank you
___
--Bandwidth and
Hi!
Yes, you can call sa many phone you wantfor example:
100,1,Dial(SIP/firstPhoneSIP/secondPhone)
makes firstPhone and secondPhone SIP phones ring at the same time when
dialing extension 100.
Giorgio
Fabio Montemaggiore wrote:
When I receive a call, only one telephone ring...
Can I
Hi all,
why a fork???
To tell the truth the Goals on the site are not clear: you can do what
you want with asterisk source.
Isn't asterisk enough?
Giorgio
Jean-Michel Hiver wrote:
harry gaillac a écrit :
Hello,
What do you think of this project www.openpbx.org ?
Something like ser and
Hi,
Junghanns drivers can be found on Junghanns site, they are named
something like BRIStuff.
Giorgio
Fabio Montemaggiore wrote:
What I can configuration my card Junghanns QuadBri?
Where I can download drivers?
Thanks?
Hi,
just do NOT type make samples: this commands writes original sounds in
your dirs even if your old custom messages are present.
(I'm talking about 1.0.7 * version, maybe in newer * versions this
command is included inside some install script).
Giorgio.
Matt wrote:
Every time I
Hi,
no, not in french, in italian but the matter is the sameI found the
only solution is to change ${VM_DATE} is to change the source code...
::))
Giorgio
Nathan Pralle wrote:
What exactly are you trying to do? Get it to say the date in French?
Nathan
gincantalupo wrote:
Hi
Hi,
I have * 1.0.7 and I have your same problem.
I dunno what version you have but till 1.0.7 simply you cannot.
We have to wait new * versions
Giorgio
amaury BOSSE wrote:
Hi all,
I don't find where you can setup the date (${VM_DATE}) in french for the mail.
Is anybody can help me?
Hi,
look insdie wiki for faxdetect instrction.
g
Rudolf Ladyzhenskii wrote:
Hi, all
Here is what I plan to do:
Have an asterisk server with 1FXS and 1 FXO port. Will have fax
machine connected to FXS and will use IP phones.
I want asterisk to detect incoming fax and swith it to fax line
1 - 100 of 108 matches
Mail list logo
