Hello, I am trying to install a Billion ISDN on Asterisk
I have Debian Etch and I installed theese packages:
apt-get install linux-headers-`uname -r`
apt-get install make
apt-get install ncurses-base ncurses-bin ncurses-term
apt-get install libncurses5 libncurses5-dev
apt-get install bison
apt-
Good morning everybody!
I have two AT-530 IP phones, when a call entry from outside (zap channel9 it
goes to 101 extension.
When I take the call and start to speak with the other person, how can I
redirect this call to 102 extension?
Thank to all.
Bye!!!
___
Hello every boy again
I have some problems with modprobe. When I type "modprobe zaphfc", this
error happens "FATAL: Module zaphfc not found."
And when I tyoe "ztcfg -vv" this error happens:
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 err
lt;[EMAIL PROTECTED]>:
We would need more details to help -- version of asterisk and zaptel
and what you did to try to install them -- hardware you have, etc and
why you did that modprobe statement.
on Thursday 05/24/2007 Josu Lazkano([EMAIL PROTECTED]) wrote
> Hello every boy again
>
>
Cohen <[EMAIL PROTECTED]>:
On Thu, May 24, 2007 at 11:17:57AM +0200, Josu Lazkano wrote:
> Hello every boy again
>
> I have some problems with modprobe. When I type "modprobe zaphfc", this
> error happens "FATAL: Module zaphfc not found."
zaphfc is part of bristu
EMAIL PROTECTED]>:
Hi Josu,
I had the same problem with wctdm.I just loaded zaptel before wctdm
and it was all ok.
Hope it can help you. :)
Giorgio Incantalupo
Josu Lazkano wrote:
> Hello every boy again
>
> I have some problems with modprobe. When I type "modprobe zaph
llion + Debian Etc), can you help?
What packages install and what steps continue.
Thanks to all and have a good day.
Josu Lazkano
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I have this card and no problem.
It is very simple to configure.
Go on!
2007/5/29, Gilles Ganault <[EMAIL PROTECTED]>:
Hello,
I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO
Bundle)
for use in a old IBM 8364 thin client:
http://www.openvox.com.cn/products_detail.php?
Hello everybody, I have installed the Billion ISDN on a Debian machine.
I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So
I connect the Billion ISDN to the ISDN Box and I call from a extension to
outside.
But it doesn't work, that is what I have in the CLI:
*CLI>
Hello everybody again.
I have a warning message in the CLI:
*CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found?
*CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found
I don't know what it me
rir Cohen <[EMAIL PROTECTED]>:
On Tue, Jun 12, 2007 at 03:54:13PM +0200, Josu Lazkano wrote:
> Hello everybody, I have installed the Billion ISDN on a Debian machine.
>
> I proved to call with a ISDN telephone conected to ISDN Box and it is
OK. So
> I connect the Billion ISDN to th
ime (1:0/1/0)
-- Executing Hangup("SIP/101-d08c", "") in new stack
== Spawn extension (SOME, 943833473, 102) exited non-zero on
'SIP/101-d08c'
Is the same, anyones know what is???
Thanks a lot.
bye!!!
2007/6/12, Tzafrir Cohen <[EMAIL PROTECTED]>:
O
Hello everybody.
I have a problem with my Billion ISDN card.
When I run Asterisk (asterisk -vvvc) on five minutes (aprox.) it puts in the
screen this:
zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff,
card = 0).
in the framelen it change 3 and 2.
Anyone knows something
where can I download that patch
thanks for respons
2007/6/13, Mauro Zanin <[EMAIL PROTECTED]>:
Try florz patch, when installing your Bristuff, for me it worked.
Ciao
Mauro
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users m
Hello everybody.
I am trying to configure an Asterisk on Debian with the Billion ISDN card. I
am using mISDN.
But when I call on the CLI apears this:
-- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new
stack
-- Called 1/943833473
P[ 1] empty_chan_in_stack: cannot empty
Hello everybody!
I have some problems with my Astersk. I have an analogical OpenVox card and
A Billion ISDN card (with mISDN).
I load the modules with modprobe zaptel and modprobe wctdm.
When I run ztcfg -vv I have this:
Zaptel Configuration
==
Channel map:
Channel 01: F
:
On Mon, Jun 18, 2007 at 12:50:23PM +0200, Josu Lazkano wrote:
> Hello everybody!
>
> I have some problems with my Astersk. I have an analogical OpenVox card
and
> A Billion ISDN card (with mISDN).
>
> I load the modules with modprobe zaptel and modprobe wctdm.
>
> Wh
The card is a OpenVox
2007/6/18, Josu Lazkano <[EMAIL PROTECTED]>:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host
Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge
00:0b.0 N
how can i see that???
thanks
2007/6/18, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Mon, Jun 18, 2007 at 02:02:53PM +0200, Josu Lazkano wrote:
> Hello, my OpneVox card is an A400P01.
>
> And the output of lspci is:
>
> 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [
Hello everybody, when I run "make config" I have this error:
install: cannot stat `init.asterisk': No such file or directory
make: *** [config] Error 1
I don't understand.
For what is "make config"? to put on /etc/init.d/?
Thanks for all
___
--Bandwi
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi configuration files are thoose:
extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten => 101,1,Dial(SIP/101,30,Ttm)
exten => 101,2,Hangup
exten => 102,1,Dial(SIP/102,30,Ttm)
exten
I can't make a "misdn show channels" on the CLI.
It looks like the mISND isn`t registered.
thanks for all
2007/6/19, Josu Lazkano <[EMAIL PROTECTED]>:
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi conf
lan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=incoming
msns=*
I don't know if the [isdn] is well
/asterisk/misdn.log.
You will have the "destination extension" as the "dad" field, IIRC.
Good luck
--
Ex Vito
On 6/20/07, Josu Lazkano <[EMAIL PROTECTED]> wrote:
> Hello everybody.
>
> I have an other problem with mISDN.
> The outgoing calls goes
Hello, the problem is nearly solved.
When I restart the computer, Asterisk load prefectly but the ISDN calls
doen't go.
I must stop the Asterisk and run "/etc/init.d/misdn-init start" and then
start Asterik.
I have a Debian machine, I need to to do something like this:
update-rc.d asterisk def
update-rc.d defaults 30 10
¿¿??
update-rc.d asterisk defaults 30 10, isn't it?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listi
thank you Tzafrir.
2007/6/21, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote:
> >
> > update-rc.d defaults 30 10
> >
>
> ¿¿??
>
> update-rc.d asterisk defaults 30 10, isn't it?
Right.
--
failed on channel 1: No such device or address (6)
The zaptel and wctdm modules are loaded correctly.
And the zapata and zaptel files are correctly too.
Thanks for all.
2007/6/21, Josu Lazkano <[EMAIL PROTECTED]>:
thank you Tzafrir.
2007/6/21, Tzafrir Cohen <[EMAIL PROTECTED]>
Hello, I have solved.
I must delete the netjetpci module from /etc/modprobe.d/blacklist:
blacklist netjetpci
thanks for all
2007/6/21, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Thu, Jun 21, 2007 at 12:48:46PM +0200, Josu Lazkano wrote:
> Hello, I have the same problem as the begini
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I want
to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten => 101,1
XX,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =>_9,2,Hangup()
exten =>_9,102,Hangup()
Or
[outgoing_trunks]
exten =>_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =>_9,2,Dial(mISDN/1/${EXTEN},45,twW)
exten =>_9XXXX,3,Hangup()
Thanks,
Steve Totaro
Josu Lazka
Hello everybody.
I have a problem with my dialplan. That my extensions.conf:
[incoming]
exten => 943712666,1,Wait(2)
exten => 943712666,2,Answer()
exten => 943712666,3,Background(/home/lazkano/welcom)
exten => 943712666,4,Wait(1)
exten => 943712666,5,Background(/home/lazkano/extension)
exten =>
Thankyou Jared, that it! it works!
2007/6/27, Jared Smith <[EMAIL PROTECTED]>:
On 6/27/07, Josu Lazkano <[EMAIL PROTECTED]> wrote:
> There is an other command to wait 4 seconds and wait for numbers?
Use the WaitExten() application instead of
Hello, I want to know if is posible to call from a mobile to my Asterisk
(national call) and then I insert an international number in my mobile and
Asterisk call with a Voipbuster account.
Is this possible?
Thanks a lot.
___
--Bandwidth and Colocation
exten => 102,2,Dial(mISDN/1/66502,30,twW)
exten => 101,1,Dial(SIP/101|30|tm)
exten => 103,1,Dial(SIP/103|30|tm)
exten => 104,1,Dial(SIP/104|30|tm)
exten => 110,1,...
I want to do that when I enter on the 110 extension.
Thanks a lot.
2007/7/25, Jared Smith <[EMAIL PROTECTED
efault,s,1 still failed so falling back to context
'default'
Apr 26 19:34:49 WARNING[3824]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1'
sent into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
The call d
59:33] NOTICE[4015]: channel.c:3066 __ast_read: Dropping
incompatible voice frame on SIP/pstn-0005 of format ulaw since our
native format has changed to 0x8 (alaw)
-- Executing [s@default:1] background("SIP/pstn-0005", "demo-instruct")
-- Playing '
ested but no musiconhold loaded.
-- SIP/103-0003 is ringing
== Spawn extension (entradas_pstn, s, 1) exited non-zero on
'SIP/pstn-0002'
Now I have a working Asterisk system at home, best regards.
--
Josu Lazkano
--
_
Hello again.
I can't configure the Billion PCI in my ISDN.
I want to know if AsteriskNow and the TrixBox LiveCDs configure it
automatically.
Thanks to all___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBS
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put "sip debug peer 101" I have this lines:
*CLI> sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI>
<-- SIP read from 10.0.0.9:5060:
INVIT
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put "sip debug peer 101" I have this lines:
*CLI> sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI>
<-- SIP read from 10.0.0.9:5060:
INVIT
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put "sip debug peer 101" I have this lines:
*CLI> sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI>
<-- SIP read from 10.0.0.9:5060:
INVIT
twork Engineer EMEA
Compuware Europe BV
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano
Lete
Sent: Tuesday, May 08, 2007 1:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] outgoing calls
hello frien
Hello again,
I have a little problem, every time I switch on the Asterisk server I must load
two modules: modprobe zaptel and modprobe wctdm
Is there any way to load there automatically when the server start?
I have a Debian Etch.
One more cuestion, it's posible to start Asterisk (asterisk -vv
Hello everybody,
I am receiving faxes and I don`t know how to receive, is there any posibility
to receive it on amail account?¿
in the console the message is this:
May 9 15:47:44 NOTICE[2618]: chan_zap.c:3703 zt_handle_dtmfup: Fax detected,
but no fax extension
-- SIP/101-0819b4f8 answere
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone calls
the Asterisk play a record (in gsm) and them the caller must choose a option
(1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose 2 it will redirect to 102 extension
if he choose
Hello everybody.
I have two AT530 telephones and one X-Lite extension conected to my Asterisk.
This is part of my extensions.con.
exten => 105,1,Answer
exten => 105,2,Background(/home/user/suport)
exten => 1,1,Dial(SIP/101,30,Ttm)
exten => 2,1,Dial(SIP/102,30,Ttm)
When I call to 105 extensi
10, 2007 10:15 AM
Subject: Re: [asterisk-users] AT530 Telephone
Hi,
What sort of DTMF do you use in the AT530 ?
It seems that just a problem of DTMF otherwise it don't work with your
softphone.
Cheers,
Josu Lazkano Lete a écrit :
Hello everybody.
I have two AT530 telephones and one
o: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, May 10, 2007 10:33 AM
Subject: Re: [asterisk-users] AT530 Telephone
Use this one
DTMF_RFC2833
Be sure to have in your peers definition this line (sip.conf):
[peer]
dtmfmode=rfc2833
Cheers,
Josu Lazka
Hello everybody!
I have a problem recording voices for my Asterisk menu.
I used the "Record(/home/lazkano/bienvenido:gsm)" function to record the menu
voices, but when I call from outside or from an extension the voice listen so
low.
is there any software to record my voice properly and conver
record voice
On Fri, 11 May 2007, Josu Lazkano Lete wrote:
Hello everybody!
I have a problem recording voices for my Asterisk menu.
I used the "Record(/home/lazkano/bienvenido:gsm)" function to record the
menu voices, but when I call from outside or from an extension the voice
liste
hello friends,
I want to install Asterisk on a Debian machine.
I need to download the sources or just with apt-get install is enought???
thanks___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or up
hello i have installed Asterisk on a Debian machine by apt-get install asterisk
I only want to call a user inside the LAN, what files I have to edit???
sip.conf???
thanks for all___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
hello I want to install Asterisk just to use in my LAN, without a analog or
digital devices.
I need to install all this packages???
Asterisk 1.2.17
Zaptel 1.2.16
Libpri 1.2.4
Addons 1.2.5
Sounds 1.2.1
thanks
___
--Bandwidth and Colocation provided b
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten => 2,1,Dial(SIP/2,30,Ttm)
exten => 2,2,Hangup
exten => 2,102,Voicemail(2)
exten => 2,103,Hangup
exten => 20100,1,Dial(SIP/20100,30,Ttm)
exten => 20100,2,Hangup
exten => 20100,10
hello,
I have two new cards, one is A400P01 from OpenVox and the other is a BILLION
ISDN.
I have Debian Etch installed.
I want install this packages:
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
http://ftp.di
hello friends, I am configurin my Billion ISDN and when I start asterisk
(asterisk -vvvc) I have this error message:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal
sign) in lin
hello, I have the A400P01 from OpenVox.
Is necesary to install all this packages?
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz
or just with asterisk
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for all.
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for all.
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for all.f
hello, I have instaled a analog line, and when I call on the console apears
that:
I want to redirect the call to 101 extension.
*CLI> -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still fai
62 matches
Mail list logo
