tech community on one of the world's most
>> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
>> ___
>> Astlinux-users mailing list
>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/list
se there is no analog signal anywhere in the chain even though the
> digital signal is downsampled as it goes through the trunk.
>
> Anyone know of a VoIP trunk vendor that will support HD voice codecs... and
> keep HD through its paths if the other end also supports HD?
>
> David
I know :(!
On Sun, Apr 16, 2017 at 4:03 PM Michael Knill <
michael.kn...@ipcsolutions.com.au> wrote:
> Too much loud music Kristian (
>
> Regards
> Michael Knill
>
> -Original Message-
> From: Kristian Kielhofner <k...@kriskinc.com>
> Rep
t
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> Astlinux-users Info Page - SourceForge
> lists.sourceforge.net
> Welcome to the AstLinux users mailing l
). Is this
still the case? Do any clients even support this?
https://issues.asterisk.org/jira/browse/ASTERISK-17856
The only option I have left is VPN really.
Regards
Michael Knill
On 16/03/2014, at 3:41 AM, Kristian Kielhofner
k...@kriskinc.comjavascript:_e(%7B%7D,'cvml','k
on the Astlinux box, then you should have the
IDS and adaptive ban enabled to stop brute force SIP registrations.
Also, use security through obscurity, and change the SIP control port to
something else other than 5060.
~Benjamin
On Sun, 2014-03-16 at 13:46 -0400, Kristian Kielhofner wrote
IAX encryption should not be trusted for anything deemed important.
Even basic reviews have uncovered several security issues and it hasn't
been well studied beyond that.
If it must be secure use SIP over strong TLS with good keys, cipher suites,
etc or use a VPN.
On Saturday, March 15, 2014,
/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
--
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--
Managing the Performance of Cloud-Based Applications
Take advantage of what the Cloud has
@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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--
Get 100% visibility into Java/.NET
@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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://goparallel.sourceforge.net
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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--
Live Security Virtual Conference
Exclusive live event will cover all the ways today's security and
threat landscape has
are graciously accepted via PayPal to
pay...@krisk.org.
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--
Live Security Virtual Conference
Exclusive live event will cover all the ways today's security and
threat landscape has changed and how
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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Exclusive live event will cover all the ways today's security
/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
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Exclusive live event will cover all the ways today's
to do that.
Sent from my iPad
Michael
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threat landscape has changed and how IT managers can
://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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Exclusive live event will cover
a username and password have
been applied and it is as follows:
username: admin
password: astlinux
The AstLinux Custom Build Engine is available immediately at the
following URL:
http://build.astlinux.org
Thanks again Lonnie!
--
Kristian Kielhofner
Can you load a net5501 with 4GB of RAM?
If not there's no reason to run a PAE capable kernel on a Soekris but it
could be the default for the geni586 image.
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Sent from a mobile device.
- Original Message -
From: Chris Abnett eldorado...@yahoo.com
To: AstLinux
.
--
Kristian Kielhofner
--
All the data continuously generated in your IT infrastructure contains a
definitive record of customers, application performance, security
threats, fraudulent activity and more. Splunk takes
}
and then define:
--
PPPOEPASS=12\$34\$abc
--
with the $'s escaped with a backslash '\'.
Unfortunately, currently, this can't be directly done in the PPPoE Password
field because a backslash '\' is treated special on a HTML POST. All
solvable though.
Lonnie
--
Kristian Kielhofner
Yes do it from the shell and it will at least work now...
On Fri, May 6, 2011 at 5:15 PM, Lonnie Abelbeck
li...@lonnie.abelbeck.com wrote:
On May 6, 2011, at 3:46 PM, Kristian Kielhofner wrote:
What about editing the file and wrapping the variable using 'single quotes'?
While that works
to support AstLinux are graciously accepted via PayPal to
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http://www.astlinux.org
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http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
from disk Asterisk will look for the
easiest file to playback. If you're using applications (MeetMe)
that require slinear (Asterisk's native format) you'll be transcoding
and now there is a big difference between alaw and GSM.
It's complicated.
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http://www.astlinux.org
http
there are already other linux developments for blackfin?
-Christopher
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..
-Christopher
I think it's about time to re-tar dialog with a version number and
host it on files.astlinux.org...
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http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
based devices always seems to leave
me wanting more.
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http://blog.krisk.org
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http://www.submityoursip.com
http://www.voalte.com
--
Throughout its 18-year
each of these had different
limitations as well...
I've had many, many positive experiences with the Nokia E series
devices (E61, E61i, E70, E71, E72, etc).
--
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http://blog.krisk.org
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http://www.submityoursip.com
http
communications, but never got around some installed limitation. I might be
mistaken and confusing this with another product/problem though.
-James
That's true too. An application can't do full duplex audio on BlackBerry.
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http://blog.krisk.org
http
..
-Christopher
I'm not sure this is even possible... Depending on the switch/DHCP
server configuration it may very well discard anything tagged with a
VLAN. Also, I don't know if you can change the adapter MAC address on
a per-vlan basis...
--
Kristian Kielhofner
http://www.astlinux.org
http
.
-Philip
With 20ms packet timing VoIP phones should generate about 50 packets
per second per RTP stream.
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port 5060
already...
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the firewall, invariably I end up using
Skype instead of Asterisk because of this.
Any ideas ?
Mart
Try OpenVPN or straight IAX2 on port 53 (usually used for DNS). I
find it is often left open.
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http://www.submityoursip.com
http://www.astlinux.org
) will have to generate our own Sangoma
config files. Depending on the complexity of the required
configuration this could be easier than it sounds or a complete
nightmare...
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http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
ace2 ace2_en phe phe_en pmm pmm_en
bogomips: 2000.80
clflush size: 64
Since I don't have any other C7 hardware to check this against, I was hoping
the list could help.
- Justin
Justin,
I have had similar results. I'm not sure why...
--
Kristian Kielhofner
http
of bytes per second processed.
type 16 bytes 64 bytes256 bytes 1024 bytes 2048 bytes
aes-128-ecb 49178.07k 182429.72k 539861.96k 1031777.26k 1214131.11k
pbx ~ #
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http://www.astlinux.org
: Compressed image found at block 0
VFS: Mounted root (ext2 filesystem).
Freeing unused kernel memory: 192k freed
Kernel panic - not syncing: Attempted to kill init!
I've tried swapping out the initrd, taking it apart, etc. No luck.
Am I alone? ;)
--
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http://blog.krisk.org
http
://astbuild.star2star.com/ast/via/
Note that this image has FreeSWITCH instead of Asterisk. It should
still boot, though :).
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http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
optimizing for
C3-2, and C7.
We should probably have a C3 generic as well... but there are damn few
of those being made.
-Philip
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http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
=073344,2,Dial(local/1...@18)
exten =18,1,Set(CALLERID(num)=${CALLERID(name)});Bug in Gateway
exten =18,n,Dial(sip/18,25,r)
exten =18,n,Answer
There probably isn't anything you can do here. The GSM carrier will
most likely not allow you to set arbitrary Caller ID numbers.
--
Kristian Kielhofner
Hello everyone,
Lonnie has officially joined the team!
I'd like to take this opportunity to welcome Lonnie and recognize
his various contributions to AstLinux. We are all very excited to
have him with us.
Thanks again Lonnie, and welcome!
--
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http://blog.krisk.org
.
For instance, I'd be willing to bet you can use a jffs2 file system.
Depending on level of compression and the build configuration, 16MB
is perfectly doable.
64MB of RAM is also possible with the proper configuration (and jffs2).
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http://blog.krisk.org
http://www.submityoursip.com
fairly substantial work on the PPP software (pppd, pppoe, etc). DHCP
is tricky enough. PPP LCP is even worse.
Philip - any comments/ideas here?
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http://www.astlinux.org
http://www.star2star.com
what the router vendors do when picking hardware, OS, etc.
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http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
://downloads.digium.com/pub/asterisk/releases
ASTERISK_DIR := $(BUILD_DIR)/asterisk-$(ASTERISK_VERSION)
Can we do this? Don't we have to support DAHDI first?
--
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http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
be cool. We
would just have to make sure the various components of the system
could use the separate user accounts in a logical manner.
For instance, the two AstLinux web interfaces practically require
root privileges. Does Lonnie or anyone else care to comment on this?
--
Kristian Kielhofner
.
Oops.
-Philip
...and wanrouter start should bring it back...
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screen?
Just better hope it comes back... ;)
Yes it's a hack but it might just work.
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http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
hardware devices hasn't
really encouraged much use of Speex. That's not saying we wouldn't
want it, I'm just pointing out that limitation.
With that being said Speex would be cool. All you should have to do
is add libspeex and make sure Asterisk 1.4 can find it and link
against it.
--
Kristian
most conferences.
Come on guys, don't let it happen to the VoIP User's Conference!! ;)
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Changelogs yet but I'm hoping newer versions
will have a more robust resolver.
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difficult upgrades could be with a tangled web of
dependencies.
With that being said - do the various chan_unistim versions *really*
depend on specific versions of Asterisk? I don't know how that could
be possible...
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http://blog.krisk.org
http://www.submityoursip.com
http
Core Not Ready (0x8)! seems to be the error.
I'm trying a build with wanpipe 3.3.14 right now but I thought I'd ask...
Any ideas?
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http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
://www.linuxtopia.org/online_books/linux_kernel/kernel_configuration/re55.html
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are /24 subnetted!
I know... That's really nasty and hidden all over the place. What's
the best way to do this when we need to - ipcalc or something?
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http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
need to be loaded by default:
ata_generic
sg
libata
sd_mod
scsi_mod
jbd
and:
sata_ for the ICH9R controller.
-Philip
Support for all of those devices *should* be compiled into the kernel
(both Runnix and AstLinux)...
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http://blog.krisk.org
http
or otherwise misbehave but you should do something
about them. Either filter TCP port 22 to the AstLinux machine or move
your SSH port to something other than 22. That will cause most of
them to go away.
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http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
need
more changing than that.
That's it. Or, you could intercept syslinux before runnix boots and type:
runnix doshell
at the syslinux command line.
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on
this as possible.
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Hello everyone,
I just ordered some VIA C7 boards and I'm looking forward to working
with them. Has anyone tried the C7 support in AstLinux yet? I'm
trying to get an idea of what I'm up against...
Thanks!
--
Kristian Kielhofner
.
Any other ideas?
Best regards,
Thorsten
Thorsten,
Thanks for the info. You need to add rootdelay=10 to your kernel
command line, not to the runnix config file. Look at KCMD=blah blah
rootdelay=10.
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be taking a while for
your USB bus to enumerate the connected devices. This delay should
allow for this and AstLinux (should) find your filesystem.
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?
-Philip
Philip,
Are the client and server behind NAT and/or a firewall? Which one
is behind AstLinux?
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Don't miss this year's
ports 21 and the entire passive port range configured
in ProFTPD - FTP Server (passive FTP)
With masquerade address and passive ports configured in Proftpd you
don't need any FTP NAT helpers. For the record, I don't like
helpers - SIP, FTP, or otherwise... :)
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on
with a clean checkout of the trunk and build on CentOS.
Thanks,
David
David,
I routinely use both CentOS and Ubuntu. Ubuntu works well when
using bash instead of dash.
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in another
thread reply
Daniel,
Try booting the trunk/0.5/0.6 versions with hpet=disable on the
kernel command line.
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Don't miss
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, is it? :)
I've attached the small patch to these e-mail to hopefully get you started.
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--- mini_httpd-1.19.bhoc/mini_httpd.c 2007-10-31 02:49:08.0 -0400
+++ mini_httpd-1.19/mini_httpd.c 2007-10-31 02:50:26.0 -0400
@@ -1638,7 +1638,6 @@
buflen = snprintf
source code anyway :).
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. Ideally you would
have a pre-configured extension range to give out to unknown devices.
That would get the phone registered and working. You could then
adjust outbound dialing permissions, etc by changing it's context,
dialplan, etc later. It would be pretty cool.
--
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On 3/27/08, Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Kristian Kielhofner wrote:
On 3/27/08, Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
I bet lighttpd would handle this more gracefully than mini_httpd does,
but not sure that it's worth the extra weight in the image
be available on files.astlinux.org, as a
last resort...
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...
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On Tue, Mar 11, 2008 at 11:51 AM, Andrea Cristofanini
[EMAIL PROTECTED] wrote:
Kristian
yes it is.
Many thanks i will check for termcap support!
Best Regards
Andrea
Andrea,
I should have been more specific - termcap/terminfo support... :)
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Hello everyone,
I have created a new mailing list:
https://lists.sourceforge.net/lists/listinfo/astlinux-devel
Astlinux-devel is for any and all queries related to the development
of AstLinux - the Astlinux Development Environment, SVN, compilation,
etc. See you there soon!
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if there are more of these with oscillator problems than
originally thought...
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card support in menuconfig, could that
be the problem ?
Thanks again to everyone here for a great project!
George
George,
Welcome!
From the log it appears that you have realtime zaptel enabled.
Disable that, make zaptel-dirclean, and try again. It should work
just fine.
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up?
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config (it
looks like logging is enabled, for one)? What firewall method (arno,
astfw) are you using? Google returns several promising looking
results for this. Have you tried any of them? What happens?
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let me know how it goes.
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it!
I have also made some consistency improvements, updated wanpipe
(fewer warnings), etc.
Thanks!
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rebuild the Kconfig, etc stuff
for you...
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filesystems. I wish there was a way to autodetect this.
Maybe we'll think of one...
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want svn export.
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On Dec 26, 2007 11:39 AM, Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Phillip,
..snip..
I've got to TRY to stop putting two ls in your name. Sorry about that!
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discovered). I have heard
that app_meetme incorporates many of the improvements found in
app_conference. If this turns out to be the case, we will certainly
remove it.
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http
serial port. Unless you add a console= line to grub you won't see
output from the kernel booting on your serial port. Try this:
console=ttyS0,19200n8
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Defy
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[EMAIL PROTECTED]
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Still grepping through log files to find problems? Stop.
Now Search log
as
being greater than 4.4. Try this:
svn up
make binutils-clean
make binutils-dirclean
rm -rf toolchain_build_i586/binutils-*
make binutils
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Still grepping
this in mind,
though:
http://www.soekris.com/Issue0004.htm
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Still grepping through log files to find problems? Stop.
Now Search log events and configuration
Gak.
Thanks,
-Philip
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/extensions.conf
Joe,
This is an Asterisk problem. You'll probably find more help on
asterisk-users.
Without seeing the full config file it's tough to say what's going
on here... I'd guess it's a SIP domain mismatch, but that's just a
guess.
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