= no trunking/meetme/etc.
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...
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of choice (sox,
Audacity, etc.) to set the desired volume level.
With sox try -V 0.25 (or -v 0.25). I can't remember if it is an
uppercase V or not.
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? Shapping based on port ranges is totally hit
or miss. TOS is the way to go:
http://www.krisk.org/files/astlinux-i586/usr/sbin/astshape
Comment out the . /etc/rc.conf and you should be okay!
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with a
fanless desktop switch.
Thanks,
Andres,
I have had people complain about the fan. I would go so far as to say
that it is very loud.
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is not in 1.2.9. I don't know what version you are
running, but either it isn't 1.2.9 or you added res_snmp on your own.
That could explain some of your other issues.
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. The real question is, why
does it need to convert to slinear?
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Kevin P. Fleming wrote:
Kristian Kielhofner wrote:
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729,
0x8140f88
Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to
find a codec translation path from g729
Kevin P. Fleming wrote:
Kristian Kielhofner wrote:
It is %100 SIP (or IAX), and I'd like to think that all of my devices
and endpoints are somewhat sane enough to properly hangup the channel to
end recording. Any thoughts?
I haven't tried it, but it appears that setting the maxsilence
' to linear format (write)
ulaw works perfectly
Now why do we need to go to slin?!?!? What magic is it trying to do
now? :)
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Douglas Garstang wrote:
Anyone know if a way to have voicemail files stored as mp3's?
Thanks,
Doug.
Doug,
I believe that asterisk-addons will add format_mp3. Also, you can just
use lame or something to convert them after the fact.
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Kevin P. Fleming wrote:
Kristian Kielhofner wrote:
Jun 1 14:55:30 WARNING[1504]: channel.c:2323 set_format: Unable to find
a codec translation path from g729 to slin
Jun 1 14:55:30 WARNING[1504]: res_musiconhold.c:299 moh_files_alloc:
Unable to set 'SIP/102c-3cd1' to linear format (write
? [1.2]
Thanks in advance,
Tamas
Tamas,
Just record your IVR prompts in g729 (or convert them to g729 from some
other format)!
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of these two points applies
to you, please sign it and provide the number of g729 licenses you have
or are looking to purchase:
http://www.petitiononline.com/ug729a/
P.S. - Anyone else who uses and loves uClibc (on gentoo, for instance)
will also benefit from this.
Thanks!
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Error LED support
- SNMP support (client, server, and traps)
- Booting from USB is back on i586
As always, you can find out more about both at http://www.astlinux.org.
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. You can
ignore this message. Better yet, configure your system to not load the
torisa.o module.
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to call your FTP upload
binary (whatever it may be) with it and force a lower transfer speed.
Let us know how it goes!
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suggestions from my previous post first! :)
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tools, which it includes:
http://www.astlinux.org/index.php?option=com_contenttask=viewid=39Itemid=38
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,
The format is a little funky, but it should work:
http://mirror.astlinux.org/sounds/en-prompts.txt
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(something like that). It's late in the day (and the week), so you'll
have to cut me some slack!
For the money, the Linksys can't be beat!
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to manually specify
a device (/dev/sda3, etc). If you are using LABEL=, try manually
specifying the device when booting.
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quality problems.
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the TigerJet
320.
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[RFC792]
8 Echo [RFC792]
11 Time Exceeded[RFC792]
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Tri,
That was last years April Fools joke. I still haven't heard this
years...
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Jeremy wrote:
Hello All:
I used the Authenticate command against a list of 4 passwords, however is
there anyway I can get these to echo in CLI for debugging purposes?
My auth line looks like this:
exten = s,2,Authenticate(/home/listofnumbers|[|a])
show application NoOp
on that architecture. I'm sure people still do
it now, it just hasn't been discussed recently.
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remember seeing some stuff in the Makefile for sparc, but that doesn't
mean that it actually works :).
As far as architectures, I have compiled it under Linux for ARM, MIPS,
PPC, and x86_64.
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that is
accessible via HTTP/FTP. That way automated build systems, etc can use
those snapshots without having to use SVN directly.
Is anyone doing this now? Where is it hosted?
Thanks.
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Tzafrir Cohen wrote:
On Wed, Mar 22, 2006 at 10:54:50AM -0500, Kristian Kielhofner wrote:
Hello everyone,
I am working on something now that could really use a snapshots.
For those that are not familiar, basically what it involves is having a
server with httpd running automatically
C F wrote:
SPA941
On 3/17/06, WipeOut [EMAIL PROTECTED] wrote:
Hi,
I am looking for a budget IP phone that can use preferably iLBC or GSM
codecs..
Suggestions?
The SPA941 doesn't support either of those codecs.
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Douglas Garstang wrote:
Looks like voip-info is down again today. *sigh*
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person uses the sound files in the OPTIONAL asterisk-sounds package.
For instance, if you were using Asterisk as a voicemail or conference
solution, you would NOT need the prompts in asterisk-sounds.
I'll (hopefully) get around to doing those someday...
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there is
no such thing)
2) Accept traffic that matches an existing state
3) Accept traffic per your requirements
4) Deny EVERYTHING
The astfw script is a good example:
http://www.krisk.org/files/astlinux-i586/usr/sbin/astfw
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Asterisk should be using these prompts. How
about a direct link this time:
http://mirror.astlinux.org/sounds/
P.S. - 2/9/2006 are the most recent.
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possible hardware!
Thanks!
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interested to find out SPECIFICALLY what happened
here. I'll be sure to keep everyone updated.
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though I like to blame MS products wherever possible, I don't
think this one was there fault. Now I'm really starting to think that
Polycom broke something in the FTP protocol on 3.1.3.
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server, web server, etc. Thanks to Asterisk, admins
of the 21st century can make a revolutionary PBX/telephone
appliance/phone switch/alarm clock/etc!
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Joseph Tanner wrote:
The PAP2 can only handle one g729 call at one time. Whether that's a
hardware limitation, or licensing, or both, I don't know.
Joseph Tanner
Hardware. The PAP2 (and SPA2000) can only do one g729 call at a time.
Any other call will have to use g711.
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/asterisk/pcom/ to reflect some new changes in this
firmware release. I just need to get my phones working first :)!
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Giordano Grandis wrote:
Hi all,
I have a simple and maybe also stupid question: if i'm in coversation on
a Zap channel and the remote party send me a DTMF, could I capture it?
Thanks all
*Giordano *
show application Read
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and make upgrade.
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-24 for such a trunk
example?
Thanks
Peter
Peter,
It think Asterisk only supports g726-32 because it is either very
similar or identical to ADPCM. Or something like that. I might be
confused.
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Jim,
Set callerid= for the IAX user/friend.
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for every participant,
regardless of what codecs the other participants are using.
app_conference might be different, but I am not sure.
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the phones register to it and trunk via IAX to the remote server.
Doug
And using AstLinux is easiest:
http://www.astlinux.org
But seriously, you should have an Asterisk server on the local side and
then do an IAX trunk (to reduce bandwidth) back to your other Asterisk
system.
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to so it will last a long time. Voicemail and
config files will be stored on a second CF that is read/write.
Did you try the development environment:
http://mirror.astlinux.org/astlinux-devel.tar.bz2
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site did not play very well with Grandstream GXP-2000
PoE. I'm betting that is a Grandstream problem (most problems are)...
As usual, my Polycom IP600 didn't have any problems with it!
I have yet to really test the other switch but I will let everyone know
once I do.
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was intrigued by your question and thought
I'd look into it. Let us know how it turns out!
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this would be the
best, most direct solution...
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Tzafrir Cohen wrote:
On Fri, Feb 17, 2006 at 11:56:52AM -0600, Kristian Kielhofner wrote:
P.S. - You need ALSA to use arecord anyways :).
What's wrond with 'rec' (from sox)? (or good old cat /dev/dsp)
Works fine with OSS and with the OSS emulation of ALSA.
Tzafrir,
arecord and ALSA
Hello everyone,
It seems that the letter s did not make it into the original release.
Please visit www.astlinux.org and download the latest tarball.Or,
if you just want s in all of the available formats, just grab this:
http://mirror.astlinux.org/sounds/s.tar.bz2
Sorry!
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Jean-Michel Hiver wrote:
Kristian Kielhofner a écrit :
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from
astlinux.org:
---
Asterisk Native Sounds are a collection
time.
In short, Asterisk will play whichever prompt is cheapest, but if it
really has to, it will play whatever it can find. It uses the file
extension to determine this. See show formats too.
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?
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.
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Brian J. Murrell wrote:
On Mon, 2006-02-06 at 11:48 -0600, Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from astlinux.org:
Which format would be best/cpu-easiest
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: 06 February 2006 17:48
To: Discussion of AstLinux - Asterisk on Compact Flash; Asterisk-
[EMAIL PROTECTED]; [EMAIL PROTECTED]
Cc: [EMAIL
Benoît Mérouze wrote:
Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from
astlinux.org:
---
Asterisk Native Sounds are a collection
for her efforts on this.
With all of those qualities, her time is not cheap. :)
Paypal to: [EMAIL PROTECTED]
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brought this up too. Basically, I'll consider
re-doing (and paying for) the sounds in asterisk-sounds based on the
donations I receive for what is provided so far in the Native Asterisk
Sounds package.
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only supports one.
The chart at:
http://www.sipura.com/products/spa2002.htm
Kind of clears it up...
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Dell systems support APIC. However, most kernels (at least
from RedHat, probably others) do not enable APIC on single processor
machines. If this is the case, try rebuilding the kernel to support
APIC on uniprocessor machines. This should resolve any IRQ sharing
problems.
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pretty sure that I have set all possible Makefile variables to
support cross compiling, but it's possible I may have missed something.
Any ideas?
Thanks!
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Innocent Evil wrote:
I prefer something 'sox' like program.
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redice.krisk.org
Do it from Asterisk!
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-w -c 1 $OUTFILE resample -ql
That will output in the proper format for Asterisk (and res_conv).
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John Reynolds wrote:
Kris,
has this module been added to Astlinux? will it be?
John R.
John,
It's in -testing. You should have gotten it when you pulled -testing
down for chan_sccp. If not, try astup testing. cat
/etc/astlinux-release should show 0.3.4.
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AMP):
[EMAIL PROTECTED] != Asterisk
AMP != Asterisk
With Asterisk, you have to edit voicemail.conf. There are some good
examples there. There is also a ton of documentation on the subject.
Did you google Asterisk voicemail, or some combination of?
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as to how these things are to be configured, it
shouldn't.
Thx.
B.
B,
I have had it working perfectly with 2.0.11 for a while now, which I
know is ancient... But hey, it works!
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, or use symlinks or some other
way to change the things that make /var grow (voicemail, CDR's, etc).
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to via a vlan
that restricts who can telnet to the phones.
Or you can move the phones to chan_sccp which lets you restart or reset
the phones from the * console.
Or you can keep using the phones with SIP and use sip_notify. I think
Ciscos support it.
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,
if people can just bring in random machines and connect them to your
network, it isn't very secure anyways :).
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SIP to SIP, with re-invites and even that is
pushing it. Overall, a Net4801 would not be appropriate for what the OP
is looking for.
P.S. - You should run AstLinux on your net4801:
http://www.astlinux.org
P.P.S. - I created AstLinux, and it rocks ;)!
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and the second priority, but
hopefully you get the idea.
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that chan_misdn is in
1.2, this shouldn't be an issue. Right?
Thanks everyone!
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!
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forms:
- Windows Install Package
- Live CD/Windows CD
- VMWare Virtual Machine Image
- Disk Image
Download now - http://www.astlinux.org
Thanks!
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=26Itemid=36
P.S. - The Windows installer has some vanity issues, but I'll fix
those soon.
P.P.S. - Of course you would not want to use this in a production
environment.
Have fun!
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QoS support
- dozens of changes brought in from -testing
ISO and CF images can be downloaded from http://www.astlinux.org. If
everything goes smoothly, 0.3 should be out by mid-week.
Thanks!
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smile, in
typical Mark style. Draw your own conclusions!
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Ben Higley wrote:
this would be very beneficial to me as well.. I have the S518 ADSL card in
my Linux system as well..
I was looking at going to an ASTLINUX solution.
Hi Kristian,
Excellent thanks..
On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Hello Everyone,
I
. I somewhat foolishly promised one by
tomorrow :).
Anyone else experiencing this? Are my CVS commands wrong? What's up?
Thanks in advance, and a HUGE thank you to everyone at Digium for
getting 1.2 out!
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Zoa wrote:
Euhm,
He was at von too, ask Kristian what a blast it would be.
Zoa.
Zoa,
I was just waiting to see how long it would take for someone to um, say
something... Why am I not surprised you were the first ? :)
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with them.
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simultaneous g729 calls. You will need
to allow ulaw/alaw on both users (in sip.conf) in case it needs to fall
back from g729.
If you need two simultaneous g729 calls, the SPA-2100 will support them.
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you need to use the files in
/asterisk/pcom/152/.
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start...
http://www.astlinux.org
P.S. - I created AstLinux, so of course I would recommend it!
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http
the same
thing. I'm sure there's something to do that doesn't involve all of that.
Matt,
In Las Vegas, I just might join you... You can pretty easily pay $12
for a drink in a club/bar!
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firmware)!
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connect
with EWSD.
(E10 and EWSD exchange store in two provinces difference. in Vietnam)
this is my logfile of E10 (successfull)
Have you tried contacting Sangoma?
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Matthew T. O'Connor wrote:
Kristian Kielhofner wrote:
Matthew T. O'Connor wrote:
Any Ideas?
Have a look at /etc/asterisk/sip_notify.conf look for:
[polycom-check-cfg]
So, from the CLI:
asterisk -r
sip notify polycom-check-cfg [name]
Isn't sip_notify.conf just an Asterisk 1.2 thing
before I
start cranking out images every 15 minutes when Kevin submits a patch! :)
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Matthew T. O'Connor wrote:
Any Ideas?
Have a look at /etc/asterisk/sip_notify.conf look for:
[polycom-check-cfg]
So, from the CLI:
asterisk -r
sip notify polycom-check-cfg [name]
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Kristian Kielhofner
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if it is
priority or not...
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Kristian Kielhofner
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strategy=ringallfree, where only members of the queue not already on a
call from that queue will receive incoming calls?
Perhaps this is something for a bounty, is anyone else interested?
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Kristian Kielhofner
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Asterisk
that they configured
the guts properly!
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they are not receiving any calls! :)
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://www.smoothfusion.com/*** ***
***t. 806.771.3873 x230***
***f. 806.771.2862**
Justin,
Check the date on that list message. It should be April 1...
April fools!!!
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