Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine,
Thomas Kenyon wrote:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
chef*CLI skype show users
Skype Users
[2009-08-02
Hi Thomas,
I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
Regards,
Emrah
Thomas Kenyon wrote:
Thomas Kenyon wrote:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0
I reported an issue on Mantis (#14).
Waiting for an update.
http://betareports.digium.com/mantis/view.php?id=14
Emrah
Emrah wrote:
Hi Thomas,
I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
Regards,
Emrah
Thomas
On 1 Aug 2009, at 22:26, Alex Balashov wrote:
Elliot Murdock wrote:
Thank you...do you know if IAX can do this?
The reason for doing is this is to get over the adsl upload/download
discrepancy. While G711 gives terrific quality, it is not always
that
feasible for the upload direction,
Tim Panton wrote:
The protocol expects the 2 ends to agree a single symmetrical codec
as part of the connection setup, but it doesn't define what actually
happens
if the codec specified in the first (full frame) voice packet isn't what
was agreed.
Asterisk only supports symmetric codec
Hi all,
I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I
want to convert them into 16 bit 8000 KHz mono so that i can use them in
Asterisk.
What is the best way of doing that?
Many thanks,
Christian
___
-- Bandwidth and
Christian wrote:
Hi all,
I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I
want to convert them into 16 bit 8000 KHz mono so that i can use them in
Asterisk.
What is the best way of doing that?
On linux you can use Sox. Google it and resd the documentation to see
how you can convert files from the command line. On windows you can
use Switch by NCH Software. Download the trial then you can pay a
small fee if you want to keep it.
Sent from my iPod
On Aug 2, 2009, at 10:30 AM,
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Sent from my iPod
On Aug 2, 2009, at 3:49 AM, Thomas Kenyon dig...@sanguinarius.co.uk
wrote:
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote:
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Could not generate Host-ID.
Make sure that you have eth0 enabled.
The MAC is used in the scheme to register and it looks like
Wireshark will support this...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Xavier Cardil
Sent: Monday, June 29, 2009 5:51 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to sniff RTP and SIP traffic only
Hi, do
On Sun, Aug 2, 2009 at 12:13 PM, randulo spamsucks2...@gmail.com wrote:
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote:
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Could not generate Host-ID.
Make sure that you
Hi,
Many thanks I used it and it worked fine.
Christian
On 2009-08-02 at 11:21 Pascal Bruno wrote:
On linux you can use Sox. Google it and resd the documentation to see
how you can convert files from the command line. On windows you can
use Switch by NCH Software. Download the trial then
Well I think thats what the problem was, I dont have it named as eth0. So
if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant
it just scan you nic handles? Can someone point me to where I can change
the NIC name in the source file or something???
On Sun, Aug 2, 2009
I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the previous version of the beta.
Hope that helps.
Tim.
On 2 Aug 2009, at 11:20, Emrah wrote:
I reported an issue on Mantis (#14).
Waiting for an update.
Hi Tim,
I don't have any skypeforasterisk process running. I tried to killall
-9 asterisk but it did not solve my issue.
Any other suggestions?
Thanks for your help,
Emrah
Tim Panton wrote:
I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the
Hello list,
Why PC modems were not used as FXO devices? Why chan_modem was deprecated?
it seemed a nicer option instead of buying expensive gateways.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13
Carlos Ruiz Diaz wrote:
Hello list,
Why PC modems were not used as FXO devices? Why chan_modem was
deprecated? it seemed a nicer option instead of buying expensive gateways.
the digium single fxo cards and clones for about $10 ARE modems.
you can get a sip gateway fxo + fxs in one box for
I did not know that the price was that low. Anyway, for people living really
far from USA the price gets incremented twice or more and this is without
considering the conversion between currencies.
1 $ = 5100 Gs., not cheap at all.
Thanks.
On Sun, Aug 2, 2009 at 3:07 PM, jon pounder
Carlos Ruiz Diaz wrote:
I did not know that the price was that low. Anyway, for people living
really far from USA the price gets incremented twice or more and this
is without considering the conversion between currencies.
1 $ = 5100 Gs., not cheap at all.
all that stuff is coming from
I have a setup with a number of customer Asterisks with T.38 enabled.
This works quite well for each customer sending faxes between branch
offices.
They all have a SIP trunk to a central Asterisk, which connects them to
the PSTN through various providers on dedicated lines. I cannot enable
I don't know then. My understanding is that the message is caused by
the wrong skypeforasterisk process running.
- did you (ever) run it as a different user ?
If it is a test box, you could try a full reboot.
Tim.
On 2 Aug 2009, at 19:35, Emrah wrote:
Hi Tim,
I don't have any
Sorry i think i forgot to mention that i have /var/spool/asterisk as a
directory from another server mounted via sshfs. when i don't use a
remote directory recording works fine. not sure if this is a permission,
but i switched to user asterisk and created new files on the remote
directory i
On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote:
Why PC modems were not used as FXO devices? Why chan_modem was
deprecated? it seemed a nicer option instead of buying expensive
gateways.
This question has been answered many times, but just for the fun of it
I'll answer it again:
If
To make your life a little easier, you can use the following filter:
sip or sdp or rtp
Just insert that into the filter query field in wireshark and it'll show you
what you need.
On Sun, Aug 2, 2009 at 12:49 PM, Joe Carroll j...@myl2n.com wrote:
Wireshark will support this…
*From:*
On Sun, Aug 02, 2009 at 03:07:08PM -0400, jon pounder wrote:
Carlos Ruiz Diaz wrote:
Hello list,
Why PC modems were not used as FXO devices? Why chan_modem was
deprecated? it seemed a nicer option instead of buying expensive gateways.
Because nobody bothered writing drivers for any of
The Astlinux Development Team is happy to announce the release of
AstLinux 0.6.7. This release is a security and bugfix release with no
new features. All current users of AstLinux are encouraged to upgrade.
Current users can upgrade either from the web interface or by issuing
the following
Greetings List,
i have a new question regarding Asterisk and E1 Cards
a client of mine is requiring an Asterisk Server with 2 E1s.
the scenario is the following
they want 400 extensions to register with the system.. and required 64
concurrent calls.
added to it that they are expecting the
On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings List,
Greetings
i have a new question regarding Asterisk and E1 Cards
a client of mine is requiring an Asterisk Server with 2 E1s.
the scenario is the following
they want 400 extensions to register with
do you suggest buying a licensed Software from Digium?
Date: Sun, 2 Aug 2009 18:53:16 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and E1 Cards
On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah tareksa...@hotmail.com wrote:
You understand perfectly fine the situation :) . I'm not saying that
Paraguay has the worse economy in South-America, but we need to work much
harder to get latest technology or to mount a tiny/small laboratory.
You will get amized if you see the things that we have done with pieces of
hardware
Yes, they changed their name to Copaco for Compania Paraguaya de
Comunicaciones. It's basically the same company ruling the whole country.
:
Oh, like ATT and Verizon here. :-(
Please pardon the editorial comment, list.
Cary Fitch
___
--
On Sun, Aug 2, 2009 at 3:30 PM, Benny Amorsenbenny+use...@amorsen.dk wrote:
Is there a way to get ONLY T.38 reinvite without Asterisk trying to get
out of the media path?
That's an excellent question. As you've realized, T.38 works by
initializing the SIP connection as audio over a chosen
The Asterisk Development Team is pleased to announce the the second release
candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of
1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for
immediate download at
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =
Asterisk Project Security Advisory - AST-2009-004
++
| Product| Asterisk|
Faxing over SIP never worked for me. The faxes would always fail. When
I saw the information about T.38, I decided to immediately upgrade to
1.6.0.11-rc2 from 1.6.0.10 and try it.
I was amazed. Without having to change anything in my configuration
faxes just worked. I have tested it with
Dear all,
I want to setup the incoming calls, that don't use authentication in
sip.conf file.
My configurations as follows,
[2000]
type=peer
host=dynamic
insecure=port,invite; (both)
context=Testing
But when I call '2000', I noticed the following message in Asterisk console,
2009/7/24 Stefan Schmidt s...@sil.at
Hello,
i´ve a question about the Meetme Options. How could i play a enter and
leave sound but without recording the user name first. I just want a
User joined conferenc and a user leaved.
With the i or I Option i have to record the name first.
Is
Jason Aarons (US) wrote:
Anyone have a customer sending/receiving multi-page faxes over Verizon
Business SIP trunk/g711 ?
Verizon Business indicates they don’t support it, and I have 2 recent
customers that it doesn’t work for, and 1 current large customer
telling me he’s going to make
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