On Mon, 2007-04-09 at 14:53 -0700, Steve Edwards wrote:
> On Mon, 9 Apr 2007, Nicholas Campion wrote:
>
> > I would much rather see the posts get ignored.
>
> Unfortunately, being ignored just invites most people to post again --
> loudly and more insistently.
>
> > This is one of the more host
On Wed, 2006-11-08 at 17:38 -0600, Eric "ManxPower" Wieling wrote:
> Maybe just allowing wildcard SIP accounts would be better. i.e.
>
> [0001547af43b-*]
>
> This would allow any device registering with any userid beginning with
> "0001547af43b-". This might be fairly easy to do.
Combined wit
On Wed, 2006-11-08 at 17:40 -0500, Peter Beckman wrote:
> So similar to a DID ringing multiple SIP extensions simultaneously:
>
> Dial([EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED])
>
> You want Asterisk, if configured, to ring all the devices that have used
> the same registra
On Tue, 2006-11-07 at 22:33 -0500, Greg Boehnlein wrote:
> Two things you need to make sure you enable when you compile asterisk:
>
> DEBUG_THREADS
> DONT_OPTIMIZE
>
> These can be toggled under the "Compiler Flags" options when you do a
> "make menuselect". Make sure that you have them turned o
On Thu, 2006-11-02 at 08:16 +0200, Stephen Davies wrote:
> On 01/11/06, John Lange <[EMAIL PROTECTED]> wrote:
> > I think Asterisk could stand to have a (much?) expanded set of internal
> > variables accessible in the dialplan.
> >
> > Just one simple example wou
On Wed, 2006-11-01 at 03:15 -0800, Luigi Rizzo wrote:
> I think (and the comments on this thread convince me even more)
> that the root of the problem is that we have one slot in the channel
> descriptor (callerid) to represent at least two (possibly more if
> we want to record the "routing" of th
Putting init.d logic in a binary? Is that what he is saying? That can't
be what he really means since the attached file is a script.
In anycase I believe whatever logic is need should be included in the
init.d/zaptel script.
On that topic I have some patches to the zaptel and asterisk start up
sc
Thanks for the clarification Slav. I appreciate the work you've done on
this. A decent JB is very important.
So, the bottom line is you set jbenable=yes on channels you want to send
dejittered audio too. In most cases this would be any channel which can
not handle its own dejitter such as Zap chan
On Wed, 2006-10-18 at 22:26 +0200, Martin Vít wrote:
> John Lange wrote:
> > I'm trying to test the new jitter buffer in 1.4beta2 but have not been
> > able tell if its working.
> >
> > In the general section of sip.conf I have the following:
> >
> > jb
ch first before implementing rate
limiting on accounts.
This exact problem has already been encountered and solved in the TCP
world (cira 2000) and the syn-cookie approach has proved itself while
connection rate limiting is known to be a poor approach.
On Fri, 2006-10-06 at 11:50 -0500, Kevin P. Fleming wrote:
> ----- John Lange <[EMAIL PROTECTED]> wrote:
> > This particular suggestion was in response to one specific type of
> > attack. At the moment Asterisk has a limit on the number of
> > authentication requests it
-09-28 at 16:40 -0500, John Lange wrote:
> Finally got our SLES 10 up and running and was dissapointed that
> zaptel-1.4.0-beta1 will not compile. Seem to be missing something in the
> kernel?
>
> Any clues as to how this can be solved?
>
> # uname -a
> Linux voip1 2.6.16.2
Finally got our SLES 10 up and running and was dissapointed that
zaptel-1.4.0-beta1 will not compile. Seem to be missing something in the
kernel?
Any clues as to how this can be solved?
# uname -a
Linux voip1 2.6.16.21-0.8-smp #1 SMP Mon Jul 3 18:25:39 UTC 2006 i686
i686 i386 GNU/Linux
# make
gr
On Thu, 2006-09-28 at 12:05 +0100, Brian Candler wrote:
> John Lange wrote:
> > A while back I posted a suggestion for limiting the impact of 1/2 open
> > SIP authentication attacks based on the principal of syncookies:
> >
> > http://lists.digium.com/pipermail/asteris
e. I think it had to do with terminating
with a "#". So:
exten => _X.#,1,
Does not work. Everything after the "." is ignored and you are forced to
rely on the digit timeout for asterisk to start doing anything. This is
exceptionally annoying in this case because the
quot;/"
exten => /[1-9][0-9]
--
John Lange
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On Tue, 2006-08-22 at 12:59 -0700, John Todd wrote:
> I stand corrected, and should pay more attention to the SVN mailing
> list. The RTCP patch was indeed committed to trunk back in June by
> Olle, and I somehow missed putting that in my notes.
>
> The values can be extracted in the dialplan,
On Tue, 2006-08-22 at 09:15 -0500, Kevin P. Fleming wrote:
> ----- John Lange <[EMAIL PROTECTED]> wrote:
> > If I remember correctly these statistics are collected in both
> > directions? The far end-point reports back statistics via SIP to the
> > server.
>
> T
I would like to second the comments about this sounding like a very
worthwhile bit of work.
Enhanced conference ability in Asterisk would be a great asset and I
look forward to seeing it added to Asterisk in the near future as it
looks like it has progressed quite far already.
John
On Fri, 2006-
On Wed, 2006-08-02 at 12:08 -0500, Eric "ManxPower" Wieling wrote:
> Sounds like ECFO.
>
> Echo Canceler Freak Out, this happens when the rxgain is too high and
> the echo canceler freaks out. Some users describe it as "screeching",
> "feedback", "static", or other useless terms. If users repo
On Wed, 2006-08-02 at 11:40 -0500, Matthew Fredrickson wrote:
> > If MG2 is the "current" canceler then shouldn't it be made the default
> > in the current release?
>
> It has had the most work done on it. However, it is in trunk, so it
> has not migrated back to a release branch yet, and possi
On Wed, 2006-08-02 at 10:43 -0500, Matthew Fredrickson wrote:
> As previously suggested, make sure you are running the most current
> version of the echo canceler (MG2 in trunk right now) before you file
> any sort of bug report. It is unlikely that you will find anybody that
> wants to fix bu
On Wed, 2006-07-19 at 20:26 +0100, Steve Kennedy wrote:
> American English, two countries seperated by a common language ;)
Ironically recorded by a Canadian.
So make that 3 countries ;)
John
___
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I guess that shows you how excited I am to finally get new sound files
for Asterisk! ;)
John
On Wed, 2006-07-19 at 13:48 -0500, Kevin P. Fleming wrote:
> ----- John Lange <[EMAIL PROTECTED]> wrote:
> > Nothing there:
> >
> > http://ftp.digium.com/pub/telephony
On Wed, 2006-07-19 at 13:33 -0500, Kevin P. Fleming wrote:
> - Strom Carlson <[EMAIL PROTECTED]> wrote:
> > "five..five.five..two.three..six..eight."
>
> This was a mistake in the packaging process. The files were not trimmed and
> padded properly; I have uploaded
I downloaded the ulaw standard prompts and dropped them into
the /var/lib/asterisk/sounds
Just hitting the voicemail system, the default seems to playback a bit
differently.
"you have no messages" is more like "you have (pause) no (pause)
messages". The spacing on the rest of the menus sounds nor
On Tue, 2006-07-18 at 08:29 +0200, Sergio García Murillo wrote:
> John Martin wrote:
> > Hi Devs,
> > I was just watching the SVN commits going by and I was wondering
> > why the H.263 buffer size in format_h263.c had to be increased to
> > 32kB?
>
> I really don't see the point if it either (
ll phones (only zero is
support in current firmware)
- Tenth byte: reserved, must be 0
- 11th and 12th bytes: multicast port number
When the caller hangs up, the server sends another multicast message
alerting the listeners that the call has ended.
---
Regar
ed in cosponsoring this feature but only if it will be
> included in the 1.4 stable release and if available for IAX as well.
>
> Best regards,
> Loic Didelot.
>
>
> On Mon, 2006-06-26 at 13:06 -0700, John Todd wrote:
> > At 8:32 PM +0200 6/26/06, Johansson Olle E
CE[27509]: app_dial.c:240 wait_for_answer: Unable to create
local channel for call forward to 'Local/[EMAIL PROTECTED]'
== Spawn extension (from-sip, 2000, 1) exited non-zero on 'SIP/2001-eece'
--
John Lange
OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web servic
direction.
John
On Sat, 2005-09-24 at 08:12 +0800, Enzo Michelangeli wrote:
> - Original Message -
> From: "John Lange" <[EMAIL PROTECTED]>
> To: "Asterisk Developers Mailing List"
> Sent: Saturday, September 24, 2005 12:11 AM
> Subject: Re: [A
aller id otherwise it
would be trivial to spoof (not that it would save you any money since
the call is free anyhow).
John Lange
On Mon, 2005-09-19 at 19:00 -0400, Joseph Benden wrote:
> This may or may not be a solution to your problem; however, I had
> problems sending toll-free calls to
y-coupled the
Asterisk monitoring will be. It may just be a host-failure detection as
mentioned but if I'm able to create or utilize a Asterisk monitoring
plug-in of some sort that would be ideal.
--
John Lange
OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Ser
cro(stdexten,SIP/${EXTEN})
As you can see, we can handle hundreds (or even thousands) of DIDs in
only a few short lines of extensions.conf. If we started assigning
device names like "joesmith" we would have to define every single
extension.
Am I missing something?
--
John Lange
Preside
m
asking.
In reality I don't really have time to hack Asterisk source code so
please let me know if the above sounds like a waste of time.
Also, if there are any developers out there that would accept a small
bounty for this work please contact me as I might go that route instead.
Thanks,
--
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