to solve it?
--
Tomasz Chmielewski
WPKG - http://wpkg.org
Software deployment with Samba
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How can I set a variable in a .call file?
I wanted to add a fax header with SpanDSP / txfax, and the information
on soft-switch.org says:
If the variable LOCALHEADERINFO has been set when txfax is run, the
value of that variable will be used as the user defined part of the
header text.
So
Alejandro Vargas schrieb:
I'm testing asteriskathome with an ISDN card
00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
I found there is the module hisax and I loaded it:
hisax 456177 0
crc_ccitt 2113 2
Alejandro Vargas schrieb:
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
possible).
I prefered to use hisax because it is already included
Jason Marshall schrieb:
I'm sure these questions have been answered at some point, but I'm too
new to this stuff to know the right words to plug into the search
function to find what I need.
I have never touched Asterisk before, but have wanted to for some time.
Now I finally think I'm going
I have the newest SpanDSP setup with asterisk 1.2.
Generally, 99% of received faxes are OK, but only about 20% of faxes
sent are delivered properly.
In zapata.conf I have set faxdetect=both, but it doesn't seem to disable
echo cancellation (I looked into asterisk logs and it says Enabled
While we are in a process of moving our office, we use soft phones which
connect over WAN/VPN to our Asterisk box in the old office.
We use IAX2 softphones configured to use iLBC.
When we call out using the softphone, the bandwidth usage is at about 3
KB/s (in and out), quality is fine.
I have some issues with sending some faxes using spandsp (receiving
faxes is generally OK).
I noticed new versions of Spandsp come out every month or two, but they
don't contain a changelog (they do, but it's outdated).
Does anyone know if one can read anywhare what changed in Spandsp?
--
Is it possible to connect to Asterisk from an external application?
What I mean, to connect and execute its own extensions, created by
some other program:
exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller)
or
exten = $NUMBER_I_WANT,1,txfax($FILE_I_WANT|caller)
and Asterisk will dial
Bharat M. Sarvan schrieb:
Hello All,
Are there any packages need to be installed before
installing the Asterisk….? Cos I am facing problems compiling the zaptel
for the Asterisk... Kindly please do let me know…
If you're starting with asterisk, you might try [EMAIL PROTECTED] -
I was wondering if there is something like that on this Earth:
Some of our users are mobile users - they are rarely in one place for
longer than 15 minutes.
They use mobile phones a lot.
From our mobile operator we have an offer which allows us to call for
free between our mobile phones.
Daniel Varella de Oliveira schrieb:
Tomasz,
I'm from Brazil, and we are using here a solution that is based on a
box where we can connect a GSM cellphone and use this directly to a
phone or PBX extension.
I think that you can use some Digium's card (FXS or FXO) on your
server, connect
Is it possible to somehow read spandsp / txfax exit codes?
What I mean, I never know if the fax sent through the Asterisk box was
sent successfully, or not (i.e., a real person picked up the phone
instead of a fax machine).
A possibility of reading an exit code, or a log file would allow to
Doug Lytle schrieb:
(...)
Is it possible to somehow read spandsp / txfax exit codes?
Run Asterisk in debug mode [asterisk -d] and use the -debug option on
the spandsp command line. Mine is as follows:
exten = s,3,rxfax(${FAXFILE}.tif,DEBUG)
After I get the debug output, I use cat
Perhaps this question should be directed to Cisco support, but since
these guys made me nuts (please check that your cable is plugged in
correctly etc.), I thought I'd ask here.
We bought a Cisco 7905G phone, which boasts to have PoE (Power over
Ethernet) support.
We have a Netgear FS108P
Sergio Chersovani schrieb:
Sergio Chersovani ha scritto:
No way to power up the phone is the the switch can be forced to send
power in any case.
I meant that the phone can power up with a custom poe injector that does
not care about 802.3af
does poe injector = poe switch (is poe switch
rulle mus schrieb:
Hello Tomasz,
I got the 7905 working with an Dell POE switch without any
modifications of cables, the 7960 also works on the Dell switch but
you have to modify the cable.
I also tried the Netgear FS108p and it does not work with the 7905,
7912 and 7960 as I have tested. Even
After reading the specifications of Cisco 7905G phone (supports SIP,
easy to manage etc.), we were so foolish and bought it.
Now we learned the hard way that we have to pay additionally for SIP
firmware.
So two months after purchase, after much struggle with Cisco
the-so-called support we
tijmen van den brink schrieb:
I set up a Cisco 7960 in about 20 minutes with this document. I hope
it works for you.
http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html
I too managed to set up a 7960 phone.
But I have (had?) a problem with 7905 phone (still
Sergio Chersovani schrieb:
Tomasz Chmielewski ha scritto:
But I have (had?) a problem with 7905 phone (still minor problems with
that, like a wrong timezone).
You can easy change it with the phone web page.
yup, I just figured that out :)
one more issue though.
any idea why a custom
Erik schrieb:
Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util
aah, now I see.
and what tool is that and where can I get this?
in my firmware package I have only two tools:
- cfgfmt.linux (a tool for converting text configuration into cisco
format, which
Sergio Chersovani schrieb:
Tomasz Chmielewski ha scritto:
I searched the whole Cisco IP Phone 7905 Series Administration
Guide, but besides the copyright notes, logo is not mentioned.
lol
http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo
I see, it's called bmp2logo.exe
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us
I've been trying to set up incoming faxes using spandsp with a HFC card.
Unfortunately, incoming faxes are of very poor quality, the pages are
not transferred wholly (sometimes only a bit of a page is transferred etc.).
From what I've read, I may have troubles with correct timing (and
need to
Tomasz Chmielewski wrote:
I've been trying to set up incoming faxes using spandsp with a HFC card.
Unfortunately, incoming faxes are of very poor quality, the pages are
not transferred wholly (sometimes only a bit of a page is transferred
etc.).
I tried sending faxes from different fax devices
Hi,
I just wanted to know if anyone managed to do call forwarding with
AT-320 phone from Atcom (not by reconfiguring Asterisk)?
For example, when I take my lunch break, I would like to forward all
calls to my mobile number.
Is it possible with Atcom AT320?
--
Tomek
I have an ISDN line with 10 numbers.
The line is then connected to * with one HFC-based card.
The format of the numbers is like below:
123456-0
123456-1
...
123456-9
Now I would like to connect those numbers to different telephones, i.e.
when someone dials 123456-0, he/she is connected to the
C F wrote:
exten = 1234560,1,Dial(phonea)
exten = 1234561,1,Dial(phoneb)
and so on...
It's called DID
OK, I had that, but it didn't work.
What I missed:
1) I have a zapata device - HFC-based ISDN card, so I needed to make:
immediate=no
in /etc/asterisk/zapata.conf (instead of
Angus Comber wrote:
I want to setup a video door entry system. I understand a lot of the
systems on the market use proprietary technology. But ideally if the
system could connect into a normal analog port or even use IP to my
Asteirsk that would be a lot better. Then I could have video
Marc Khayat wrote:
Hello all,
I just installed Asterisk 1.0.7 and astguiclient so you can say Im
very new at this.
How can I manage my Asterisk using the web or somehow, since there are
too many configuration files and too many variables
You may take a look at [EMAIL PROTECTED] - it includes
Matthew Boehm wrote:
Before you jump ahead, yes I do have chan_zap.so loaded..
Call Flow: Asterisk 1 --IAX2-- Asterisk 2 --- PRI
-- Accepting AUTHENTICATED call from 22.22.22.22:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs =
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel
Matthew Boehm wrote:
Tomasz Chmielewski wrote:
Matthew Boehm wrote:
snip
first, let me know, if you can dial yourself? (i.e. PSTN - * - that
(zap) card)
Yep. I sure can.
so everything seems OK.
I guess we would need:
- /etc/zaptel.conf
- /etc/asterisk/zapata.conf
- the construction
Kib Eki wrote:
Hi,
can one summarize the main differences between Asterisk and
[EMAIL PROTECTED] or
point me to a location where i can find such a list?
See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk
site - www.asterisk.org.
Basically, asterisk is a program, and [EMAIL
I have a HFC-PCI based ISDN card.
How should an extension be constructed, when I want to set up a specific
outgoing number (I have 10 or so MSN numbers)?
For example, when I call 6546 from my SIP phone, I would like to call
100 with an outgoing number of 555 - how should I do this?
exten =
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I can
call myself and talk to asterisk :)
I have defined an extension:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
exten = _0.,2,Congestion
exten = _0.,3,Hangup
So
Deti Fliegl wrote:
Tomasz Chmielewski wrote:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
set g0 instead of 0:
exten = _0.,1,Dial(Zap/g0/${EXTEN:1})
Yes, it changed something, now I get an immediate hangup:
-- Executing Dial(SIP/201-e124, Zap/g0/98) in new stack
-- Called g0/98
-- Channel 0/1
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at 1
Remco Barende wrote:
Then when I try to start asterisk I get this error:
chan_zap.so: load_module failed, returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
May 2 20:41:58 WARNING[8663]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
[EMAIL
pinchien wrote:
What is Asterisk GUI architecture acturally? I could not get it...
hmm?
check [EMAIL PROTECTED] - it contains AMP - http://asteriskathome.sf.net
Tomek
--
Startuj z INTERIA.PL! http://link.interia.pl/f186c
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
I also got a diva pci 2.02 card, but although the kernel sees the
incoming calls, asterisk refuses to answer. Did you have this issue at
all?
The kernel seems to be denying the call...
if you see the calling in the systlog, that's 98% of success :)
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
Previously I did get asterisk to see the call, but not currently.
This is in the usa, so my msn is a 7 digit number.
The kernel is saying the following:
Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 -
2781980
so this is your
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
When I call my second msdn, I get the following:
== Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so
falling back to exten 's'
-- Executing Answer(Modem[i4l]/ttyI1, ) in new stack
somersvoip*CLI Apr 28 04:04:33 localhost kernel:
I'm trying to configure an asterisk box with two cards.
Incoming calls are working fine with two ISDN cards, however, I am able
to make outgoing calls only through the first card.
exten = _0.,1,Dial(Modem/g1:${EXTEN:1})
exten = _9.,1,Dial(Modem/g2:${EXTEN:1})
If I try to use the second card,
Can asterisk send AT commands to a modem?
If so, how?
I have two ISDN cards (with i4l - capi4linux doesn't work with them),
and would like to specify which card to choose for dialing out (without
it, i4l uses first free /dev/ttyI device).
Tomek
___
Amit Singla wrote:
Hi Everyone,
I have bought a Digium TDM400P and am using asterisk on RedHat 9.0. I
was able to configure Asterisk and SJPhone, so I have been able to call
from IP to IP and also from IP to a analog phone which is attached to
the digium card.
My problem now is to dial from an
I'm new to asterisk and still learning it.
I wanted to ease my efforts a bit and use AMP (Asterisk Management
Portal), and see what changed in the config files when I use it.
However, I realized that I can only add SIP, IAX2 and ZAP extensions - I
didn't see an option to configure an ISDN
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten =
Umair Bari wrote:
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
no, still does not hang up :(
I have to pick up the phone and hang up manually (or kill
I just wanted to let you know that it's possible to use Eicon DIVA PCI
2.01 ISDN cards (not server divas) with asterisk.
First thing to do is to load the module. If you have two of these cards,
you should do it like that:
modprobe -v hisax protocol=2,2 type=11,11
And now you can have up to 4
Corvin wrote:
Hello!
I've encauntered some serious problems with asterisk.
I have to install it on system:
1. Mandrake 10.1
2. kernel 2.8.1
3. four ISDN cards.
And I am in big trouble,
isdn4linux is no longer supported for kernels 2.6 (on this system there are
not any /dev/ttyI0 and similar
Hello,
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
I started asterisk from
Hello,
Another warning I have.
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
I
Hello,
Yet another warning I have.
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
Hello,
I just sent it with a wrong title... so once again:
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't
Hello,
Is there a list of software phones which will work with Asterisk?
For Linux and Windows?
I don't have any hardware yet, and before I buy anything I would like to
know how Asterisk really works (with software phones for example).
Tomek
___
Patrick wrote:
On Wed, 2004-12-01 at 11:26 +0100, Tomasz Chmielewski wrote:
[snip]
chan_iax.c:7507 load_module: Unable to open IAX timing interface: No
such file or directory
What does it mean? Is it something to worry about? How to get rid of it?
For these and many other basic questions first
Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still getting the same error.
Had you actually
Peter Svensson wrote:
On Wed, 1 Dec 2004, Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still
Dave Cotton wrote:
On Wed, 2004-12-01 at 13:37 +0100, Dave Cotton wrote:
On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote:
So the only issue left I have is with this skinny not found when 0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
Roger Hanson wrote:
- Original Message - From: Tomasz Chmielewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 4:42 AM
Subject: [Asterisk-Users] software phones for Asterisk - is there a list?
Hello
Hello,
I'm trying to get my Eicon Diva 2.01 PCI ISDN card working with Asterisk
and was told that I still have some chance if I tried using chan_capi.
I tried compiling it (chan_capi) on two systems running 2.6 kernels, but
got lots of errors.
Before I go investigating - is it possible to
Hello,
I can't compile chan_capi-0.3.5 (also tried with 0.3.4b).
I tried compiling it on two systems with very similar (unsuccessful)
results:
1) SuSE 9.1 on 2.6.5 kernel,
2) Mandrake 10.1 with kernels 2.6.8.1 and 2.4.27, using gcc 3.3.4 and
gcc 3.4.1
I'm getting the following errors (similar
Derek Conniffe wrote:
Chan_capi works fine on a 2.6 kernel for me (2.6.8 and SuSE 9.1) but I'm
using a AVM Fritz PCI V2 BRI card and I firstly installed the AVM fritz PCI
capi driver and then I installed chan_capi - everything went very smoothly
and I've installed with different AVM Fritz cards a
Jean-Michel Hiver wrote:
Tomasz Chmielewski wrote:
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my
PC-Asterisk server (which I yet have to install) and use them as 4
lines in case
Patrick wrote:
On Mon, 2004-11-29 at 13:51 +0100, Tomasz Chmielewski wrote:
[snip]
It appears they have been wrong.
I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI
2.0... This same page says this card is is not Linux compatibile, though, but
it is.
Capi.org
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my PC-Asterisk
server (which I yet have to install) and use them as 4 lines in case
anyone has to call me in / I have to call out using
Jean-Michel Hiver wrote:
Tomasz Chmielewski wrote:
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my
PC-Asterisk server (which I yet have to install) and use them as 4
lines in case
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