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Hi all!
This is my first message to the list/newsgroup.
This weekend and after to have fought by some time with my soundcard
with respecto to the voice capture, after assuring to have solved that
problem, I installed Asterisk on Debian GNU/Linux
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Hello Daniel,
Hi Dana.
You will find the information at http://www.voip-info.org/ and
http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the
Online Book link) very useful.
I have the second edition that covers Asterisk 1.4 and
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El domingo 10 de mayo del 2009 a las 17:12:51 -0300,
Daniel Bareiro escribió:
I suggest testing your SIP softphone with the Echo() and/or
Playback() dialplan applications before attempting to call another
softphone/hardphone/etc. This will allow
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Hi all!
Days ago I bought a OpenVox A400P card with a port FXS and another FXO
that I am testing with my Asterisk installation in my house. I'm using
Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on
Debian GNU/Linux Lenny.
I was
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Hi Tzafrir.
El miércoles 20 de mayo del 2009 a las 10:00:46 -0300,
Tzafrir Cohen escribió:
On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
Hint: you don't need to set 'signalling' for analog channels. Or just
set it explicitly
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Hi Dave.
El miércoles 20 de mayo del 2009 a las 18:12:04 -0300,
Dave Fullerton escribió:
I load the modules wctdm and dahdi. But when I execute in Asterisk
CLI dahdi show channels, I get the following error message:
No such command 'dahdi show
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El miércoles 20 de mayo del 2009 a las 21:19:18 -0300,
Daniel Bareiro escribió:
I load the modules wctdm and dahdi. But when I execute in Asterisk
CLI dahdi show channels, I get the following error message:
No such command 'dahdi show channels
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Hi Tzafrir, Danny.
El jueves 21 de mayo del 2009 a las 06:55:14 -0300,
Tzafrir Cohen escribió:
Mmmm... but I believe that it had done already in that order. In fact, I
reviewed the existence of the module and it was in the directory. For that
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Hi Tzafrir.
El domingo 24 de mayo del 2009 a las 17:33:36 -0300,
Tzafrir Cohen escribió:
# cat /proc/dahdi/*
Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
1 WCTDM/4/0 RED
2 WCTDM/4/1
3 WCTDM/4/2
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El domingo 24 de mayo del 2009 a las 19:38:30 -0300,
Daniel Bareiro escribió:
Now it would remain to find the cause of why I cannot call from a SIP
extension to an analog telephone. Perhaps it is by something related
to the contexts
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Hi all!
Making some changes in extensions.conf to test the incoming calls so that
these are derived to a SIP extension, I found something that draws attention
to me: if I test calling to my PSTN line from a mobile phone, when take the
call from the
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Hi all!
I'm trying to doing a transfer from an analog extension to a SIP
extension but until the moment I was not successful.
I was testing both the recall key and uncomment the following
lines in the features.conf file:
blindxfer = #1
atxfer = *2
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Hi Tilghman and Grygoriy.
Tilghman Lesher escribió:
I was testing both the recall key and uncomment the following lines
in the features.conf file:
blindxfer = #1
atxfer = *2
verifying previously that the extension uses the arguments tT with
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Daniel Bareiro wrote:
As I've commented in a previous message, after dial *60 (of *600 to Echo
test), I obtain like a tone cut in three parts followed of a continuous tone,
causing that I'm incapable to dial the extension completely
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Hi all!
Now that I have a little more time, I was debugging my dialplan and it
was of the following way:
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; DGB - 20090615
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten =
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Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.
/etc/asterisk/sip.conf:
; Outgoing to ekiga.net
[ekiga]
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SIP wrote:
Daniel,
Hi SIP.
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't believe
that this solves what I
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SIP wrote:
Daniel,
Hi SIP.
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't
believe that this solves what I
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El miércoles 19 de agosto del 2009 a las 08:04:17 -0300,
SIP escribió:
Daniel,
Hi SIP.
I'm a little confused as to what I'm seeing here. You're bounding
through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is
this some sort of
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Hi all!
I'm investigating the possibility of using Asterisk as much for internal
communication in an office as between offices and I would like to know
what considerations you could comment to me being based on the
experience that you have had.
A
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Hi, all!
I'm being based on this document [1] to send and to receive calls using
ekiga.net. But I'm seeing, in an Asterisk console, several messages of
this type:
[Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout:--
Registration
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Hi, Warren.
On Thursday, Feb 18, 2010 at 00:01:23 -0300, Warren Selby wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost instead
of externip?
It can
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Hi, Warren.
On Thursday, Feb 18, 2010 at 16:30:40 -0300, Daniel Bareiro wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost
instead of externip?
It can
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On Thursday, Feb 18, 2010 at 17:29:44 -0300, Daniel Bareiro wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost
instead of externip?
It can be. I
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On Thursday, Feb 18, 2010 at 05:36:41 -0300, Administrator TOOTAI wrote:
Hi
Hi, Daniel.
Daniel Bareiro a écrit :
[...]
Hours ago the IP changed and the domain was updated satisfactorily,
but in spite of this I was obtaining the registering
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Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI
-- Executing [...@from-internal:1]
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Hi all!
I'm testing with a Grandstream BT200 telephone and, according to I read,
it has a LED that blinks if for that extension messages were left.
In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the
extension in which my Asterisk
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Hi, Gordon.
On Sun, 21 Mar 2010, Gordon Henderson wrote:
I'm testing with a Grandstream BT200 telephone and, according to I
read, it has a LED that blinks if for that extension messages were
left.
In Voice Mail UserID, under the ACCOUNT tab, I
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Hi, Alyed.
On Mon, 22 Mar 2010, Alyed wrote:
I was with the following situation: if I call from a cell phone, my
Asterisk take the call, it presents to the caller the possibility to
dialing an extension number and, in case of not doing it, it
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Hi, Alyed.
On Mon, 22 Mar 2010, Alyed wrote:
you are right, under [channels] is where it's supposed to be my
mistake, i guess i was thinking in sip.conf :)
Perfect :-)
However, the following doubt arises to me: it would also have had
this
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Hi all!
I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and
dahdi-tools-2.1.0.2 compiled by myself with the source code of the
official site of the project. I would like to update to one more newer
version. I suppose that the recommendable thing
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Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
Make sure not to do make samples or you will overwrite your .conf
file. This is the important one to watch out for. You can save off
your .conf files and then restore them or compare your files
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Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
Make sure not to do make samples or you will overwrite your .conf
file. This is the important one to watch out for. You can save off
your .conf
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On Sun, 28 Mar 2010, Alyed wrote:
My idea is to continue making the configurations by hand at the
moment, that it is the way that I used until now, to familiarize to
me with the handling of Asterisk at lower level, without using a
graphical
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Hi, Alyed.
On Sun, 28 Mar 2010, Alyed wrote:
I didn't know that there was Digium's GUI. It is FLOSS? I was looking
for in the site of Digium in the download section, but the unique
thing that I saw that it speaks of a GUI is AsteriskNow, that in
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Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
I think if you are installing dahdi complete from source you do
make all and make install and make config
Something that I forgot to ask previously is if the update of Asterisk
or DAHDI is
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Hi all!
I'm trying to test with a friend who has an Asterisk in his office with
the Asterisk which I have in my house. Then I have an extension that he
is trying to register remotely.
Trying with the Twinkle client, I see that it is registered:
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Hi, Alyed.
On Sun, 11 Apr 2010, Alyed wrote:
Daniel, you are having a problem often seen in pre 1.4.14 versions.
Before this release srvlookup=no was the default for sip.conf and
guess the same for iax.conf . So if you are working with a
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Hi all!
In the network of my house I was testing the security with my Asterisk
installation. The first test that I'm doing is an man in the middle
attack.
In this scenary, the attacker is a virtual machine that it tries to see
the SIP traffic
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El jueves 22 de abril del 2010 a las 14:33:01 -0300,
Philipp von Klitzing escribió:
Hi!
Hi, Philipp.
But it draws attention to me between the PC with softphone and the
telephone I see traffic ARP or ICMP that could make to try between
the
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Hi, Steve.
On Fri, Apr 23, 2010 at 22:38:49 -0300, Steve Totaro wrote:
Perhaps it was not very clear, but yes, I was talking about this. I
believe that I found the cause of the problem. The cause by which I
was not seeing VoIP traffic between
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Hi all!
During tests with a Grandstream GXP280 phone, I found that pressing
'hold' button, the other extension (Qutecom softphone) is put on hold
but without music. Then, when resuming the conversation, I listen the
other user again but he/her no
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Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf is the following one:
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Hi, Tzafrir.
On Thu, May 20, 2010 at 09:58:26 -0300, Tzafrir Cohen wrote:
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration
that I'm using in chan_dahdi.conf is
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Hi all!
I had the opportunity to test a Sangoma A200 card and I have some doubts
that I would like to consult:
I tried to detect the card and I had no success using the wctdm module
with DAHDI. I guess this is because electronics is different
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On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
Greetings!
Hi, Tim!
I had the opportunity to test a Sangoma A200 card and I have some
doubts that I would like to consult:
I tried to detect the card and I had no success using the
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Hi, Gopalakrishnan.
On Fri, May 28, 2010 at 01:44:41 -0300, Gopalakrishnan A.N wrote:
I suspect the channel is not ceased correctly in Siemens PBX, since
you get dial tone from Siemens PBX the channel from Asterisk is
rejected in your Siemens
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Hi, John.
On Fri, May 21, 2010 at 23:35:41 -0300, John Novack wrote:
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa. For
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On Wed, Jun 02, 2010 at 21:50:43 -0300, Daniel Bareiro wrote:
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa
Hi all!
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against the
local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
found that it can be related to a bug of chan_sip, can
Hi, Phil.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
found that it can be related to a bug of
Does you Asterisk server point to an internal DNS or to your
router ?
The /etc/resolv.conf of the host on which I installed Asterisk
points to an internal DNS. Is there a parameter in the Asterisk
configuration where also I have to force the use of an internal
DNS server?
Do your
Hi, Alejandro.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
You have to be a bit more specific. For example is your Asterisk box
behind a router/nat? Or does your
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Hi all!
Some time ago I'm using Asterisk (currently 1.8.10.0) at home to manage
the calls. Nothing yet very complex, just something compiled by me using
the source code from the official site of the project and configuring
the files manually to both
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