On Thu, Jun 16, 2005 at 03:27:49PM +0100, Alistair Cunningham wrote:
I'm planning an Asterisk Voicemail system of around 3000 users spread
across several sites, each site connected by a fast network to a central
site. We're considering 2 models:
- Central Voicemail with VoIP calls from
On Mon, Jun 13, 2005 at 12:48:46PM -0700, Robert Hajime Lanning wrote:
quote who=trixter http://www.0xdecafbad.com;
Protecting freedoms by putting limits on (thus restricting freedoms).
Interesting concept.
It maybe an interesting concept, but it is absolutely true.
True anarchy (no
it's simply #2. They are taking HEAD and maintaining a version where
they are extraordinarily careful about what goes in. Similar to what
stable was supposed to be.
So is there at least a cvs tag? Can I cvs co -r ABE asterisk?
-w
--
William Waites, Consulting Technologist
Consultants Ars
On Mon, Jun 06, 2005 at 12:20:13PM -0400, Andrew Kohlsmith wrote:
On Monday 06 June 2005 11:25, William Waites wrote:
So is there at least a cvs tag? Can I cvs co -r ABE asterisk?
Honestly, what part of the source is not available do you have trouble
comprehending?
Sorry, due to the high
On Mon, Jun 06, 2005 at 03:31:31PM -0400, Andrew Kohlsmith wrote:
So Digium has leveraged the community to build for them a
proprietary product. Correct?
Nice.
Others have commented on this, so I'll refrain short of saying you need some
serious clue. I'm not sure I see your name in
On Mon, Jun 06, 2005 at 05:11:42PM -0400, William Waites wrote:
If you're interested, take a closer look. chan_sip.c, some time ago.
Miscellaneous bug fixes. But a whole lot, and not for a long time.
^^
should read not a whole lot. argh.
73
-w
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On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote:
Is there a traceroute equivalent in the VoIP world? I would like to see the
route a call takes after it gets to the gateway. Basically showing all the
hops until it reaches it's destination or PSTN termination.
For SIP, there is a
On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote:
Is there a traceroute equivalent in the VoIP world? I would like to see the
route a call takes after it gets to the gateway. Basically showing all the
hops until it reaches it's destination or PSTN termination.
Note that sipsak
On Wed, Mar 10, 2004 at 05:01:38PM -0500, Jeremy McNamara wrote:
That fact is not the problem. It the fact that there is no FORK of
Asterisk that Digium secretly maintains. This is how rumors get
started.
If memory serves, you were the one who started that rumour.
I remember you claiming
Jeremy, I am really not interested in rehashing this again.
You know my views on the matter, I know yours. We disagree.
On Wed, Mar 10, 2004 at 08:04:52PM -0500, Jeremy McNamara wrote:
I seem to recall your http://www.gnutel.com publicly discussing a fork
of Asterisk.
This is no secret
On Thu, Mar 04, 2004 at 11:24:12AM -0600, Tilghman Lesher wrote:
You've already answered your question. As chan_h323 does not
work on FreeBSD, and as you need chan_h323, you are therefore
required to not use FreeBSD.
Install Linux, like everybody else.
Genetic diversity in operating
On Thu, Mar 04, 2004 at 02:49:52PM -0600, Steven Critchfield wrote:
Genetic diversity in operating system support is a good
thing. It makes for more robust code. Following standards
is a good thing -- POSIX was written for a reason. If you
only support one OS you are less likely to
On Mon, Mar 01, 2004 at 05:32:34PM +, WipeOut wrote:
Currently connecting more than 3 analog lines to asterisk can be
problematic unless you get hold of a channelbank (not that availible in
the UK)..
Of course there is a 12 port configurable FXS/FXO
blade from VoiceTronix
/w
--
On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote:
Thanks William, it's get.
but new problem:
snip
server dont have any sound device ( I think:) )
Why noone make normal Makefile and FAQ for FreeBSD Asterisk.. als many for
Linux.
I don't know why the Asterisk crowd is resistant to
On Sun, Feb 29, 2004 at 04:26:17AM +0100, Serge wrote:
Hello,
Pls. help !
I have server on Freebsd 5.2 and don't may install asterisk , following errors: (
gmake clean ; gmake install )
-
include/mpool.h:53: error: syntax error before
On Wed, Feb 11, 2004 at 02:02:03PM +0100, dkwok wrote:
-- Format for call is G729A
^
I suspect that if you use a standard
format your call will go through. Also
keep in mind that there is no reason
to go through IAXTel for this -- it is
just necessary to dial
On Mon, Feb 09, 2004 at 02:28:02PM +0100, Dawid Mielnik wrote:
Would the t.38 transmission be properly handled by the t.38 supporting end
points whith mediastrem passing through Asterisk ? (dont have much
experience with t.38) Has anyone ever tried anything similar / different /
wierder to
On Mon, Feb 09, 2004 at 09:31:30PM +0100, Philipp von Klitzing wrote:
Why exactly would hylafax be a worst case solution only, why would you
tink that that the Asterisk solution is better at all?
The worst case would be the modem hairpinned into an FXS
port, not hylafax per se.
Instead
On Wed, Feb 04, 2004 at 10:18:10AM +0100, Andy Powell wrote:
but apparently this will never make it into CVS
(since the engine is not GPL)...
GPL code is not allowed in the Digium CVS repository.
Only split-licensed code is.
/w
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On Tue, Feb 03, 2004 at 01:04:27PM -0500, Matthew B Marlowe wrote:
exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)
The music on hold will not work
I believe you do not want a comma between the t and the m.
-w
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On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote:
This thread got me thinking of other servers that would run asterisk. The
obvious question comes up if Xebian (the xbox version of Debian) would run
as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.
I
On Sun, Feb 01, 2004 at 04:51:30PM -0600, Steven Critchfield wrote:
This isn't intended as a flame bait. The original message should have
been more clear that I thought you where experiencing crap in windows.
Heh. I haven't used windows since 1995 :)
In fact, with HP-UX you cannot delete or
While your problem is most likely bad RAM as other
replies have suggested, there is another thing to
keep in mind.
Some implementations of dynamic module loading have
problems if a loaded module is overwritten on the
disk. What this means is that it is safest to stop
Asterisk just before
On Thu, Jan 29, 2004 at 06:27:33AM -0600, Rich Adamson wrote:
We are thinking of making network of about 25000 extension numbers.
These extension will be SIP phones. Asterisk will be connected to some VoIP
gateways through H323 which will allow to
terminate calls.
Can Asterisk handle
FYI and to whom it may concern, I have made Debian
packages of Asterisk et. al. You still need to build
a new kernel and the zaptel modules from source, but
Asterisk and libpri are manageable with dpkg.
The debs as well as mirrors of the source distribution
are here:
On Fri, Jan 23, 2004 at 08:45:07AM -0800, Kostur, Andre wrote:
v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not
on an i386)
Ah, I didn't realize 0.7.1 was in unstable -- I run
mostly testing here.
What do you have different in your packages?
Nothing in
On Tue, Jan 20, 2004 at 03:09:54PM -0600, Tilghman Lesher wrote:
The specific issue is that VoiceAge uses a copy protection method
that binds the license to the filesystem.
Solution: don't use proprietary software. Then you don't have to
worry about the stupid things that they do to keep
On Wed, Dec 17, 2003 at 04:45:14PM -0500, C. Maj wrote:
My nose is bleeding from CVS. Same thing with a
T400, had to comment out all fax extensions. Updated
to CVS of 12/16.
We really need to get this organized.
0.5.0 is too old to be useful, and having people run
CVS snapshots in
On Tue, Dec 16, 2003 at 10:59:50PM -0600, Walker Haddock wrote:
edit iax2.h file and change line 73 as follows:
#define IAX_DEFAULT_PORTNO 80/* 4569 */
this is *really* the *wrong* way to fix it.
the correct way is to set port = 80 in iax.conf
BUT...
you will notice near
On Mon, Dec 15, 2003 at 10:05:56AM +0200, Peter Zeltins wrote:
My Asterisk box also does NAT for internal network, and
establishes site-to-site VPN tunnel(s). As a result I have
several internal interfaces with private addresses on them, and
only one public interface. By trial-and-error I've
On Mon, Dec 08, 2003 at 11:05:17AM -0600, Steve Dolloff wrote:
Local server:
register = [EMAIL PROTECTED]
;
[voip2p]
type=peer
host=dynamic
port=4569
trunk=no
qualify=yes
context=IAX
Remote server:
register = [EMAIL PROTECTED]
;
[voip1p]
type=peer
host=dynamic
port=4569
On Mon, Dec 08, 2003 at 10:27:15AM -0600, Dave Weis wrote:
What about McLeod USA?
They aren't necessarily evil, just incompetent.
Any sufficiently advanced incompetence is indistinguishable
from malice
-- Jamie Reid
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On Mon, Dec 08, 2003 at 07:46:50PM -, David J Carter wrote:
Hi,
I have chan_sip.c version 1.259 do I still need the patch.
yes.
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On Thu, Dec 04, 2003 at 10:35:13PM -0800, Andrew Gillham wrote:
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /usr/src.
i have not gotten the zaptel drivers to
On Fri, Dec 05, 2003 at 11:58:44AM -0600, Andy Hester wrote:
The guy did leave open the possibility that he could be wrong, and said that
he'd be glad to answer any further questions or if we had some other way of
doing it. If you or some of the others think that this should be possible
On Thu, Dec 04, 2003 at 07:56:56PM +0100, robert ivanc wrote:
this patch seems to break my GS phones that are connecting to * via NAT.
The one before that works ok - 249 or something? They can't connect
anymore - get a Not Found error back.
That is very strange -- the *only* difference
On Thu, Dec 04, 2003 at 04:58:03PM -0600, Steven Critchfield wrote:
a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
over the bus 10 times, you're still only using up half the
peak bandwidth.
Thats only if you
On Thu, Dec 04, 2003 at 03:31:06PM -0800, David Boreham wrote:
There are DS3 (and OC-3) PCI cards available
with Linux drivers (for data). Might be worthwhile
contacting a vendor of those things to see if there's
a way to suck the TDM voice data
off a channelized DS3.
I know of OC3 ATM
On Thu, Dec 04, 2003 at 06:25:16PM -0500, William Waites wrote:
btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386
doing well in excess of 500Mbps so it /is/ possible.
Just another data point:
We also made measurements in November 2000 from a Pentium III running
Linux
On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
What's the correct way to do cvs update now?
'cvs update' seems to work in the asterisk directory, but not the zapata
or other source directories.
I use 'cvs update -PAd'
AFAIK it should work in the zapata and libpri
On Wed, Dec 03, 2003 at 05:47:59PM -0200, listas iPfone wrote:
Hi!
I need help to undestand the options:
hello.
externip= static/ dynamic ip? can be a domain?
externip can by an IP address or a domain. it uses gethostbyname(3)
in the code.
localnet= internal ip of * machine?
Ok, so you've read the Wiki and gotten call routing using ENUM to work
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing)
with your own ENUM-alike domain, e164.example.com.
But how do you populate it with data? You can do it manually, but that gets
very tedious very
Query: why does Digium not make a 2 or 4 port FXO interface card?
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On Mon, 3 Nov 2003 17:10:10 -0600 (CST), Martin Pycko wrote
It doesn't care about the phones. If you phones are behind nat use nat=yes
for each defined account.
The fix is incorrect. Asterisk chan_sip.c must distinguish between
SIP peers that are behind the firewall (together with the *) and
On Tue, 4 Nov 2003 13:00:44 +1100, Anthony Wood wrote
Internals can use the IP address of the NAT box as the Asterisk
Server IP and then it should work.
i.e. don't set your internal SIP UAs to connect to the internal IP
address of the Asterisk Server.
The fix allows asterisk to work
On Mon, 3 Nov 2003 19:33:36 -0800 (PST), Chris Albertson wrote
Did your patch make it to CVS? Sorry for being lazy and not looking.
From the sounds of thing maybe only half the patch made it
But I'm not at the right machine to look at present.
No, but it may have something to do with the
On Fri, 31 Oct 2003 09:09:22 -0800 (PST), Chris Albertson wrote
Stephens, I think preferably, introduces one new sip.conf
line for the internal _network_ which acceprts a network
address in the form inside=192.168.111.0/14 Where the 14
would be the number of zero bits in a 32-bit mask
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