14 dec 2007 kl. 11.20 skrev Andres Gomez:
Hello List
I am very interested in developing a research project on security
protocol for VoIP, under the GPL.
For some time I have been reviewing ZRTP, I would like to know the
opinion having regard to whether and under asterisk, but I see
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
I realize that 1.4 at release time wasn't ready for release, but we've
spent one year
Hi,
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
Just my 2 cents
I have more than 70 running servers installed with 1.2, we
Hello everybody,
Since 1.4 release our company installed more then 200 Asterisk servers using
Asterisk 1.4 version.
At start we had several bugs with SIP channel and CDR handling but starting
from 1.4.6 or something it works without problems.
We are really happy with 1.4 and thank you for
Hi
i've installed this software:
SOFTWARE
mISDN-1_1_7
mISDNuser-1_1_7
Asterisk-1.4.15
SOFTWARE
misdn is correctly loaded by misdn-inist start
Here there is the misdn.conf (copied from an existing and working
installation with Asterisk 1.2.x and one
Hi,
Does anybody know where I can find any open source ITU G.107 implementation
available? I'm looking a way to measure the voice quality in my projects..
Thanks in Advanced,
My Best Regards,
Andre Lomonaco
___
--Bandwidth
Hi Olle
2007/12/15, Olle E Johansson [EMAIL PROTECTED]:
14 dec 2007 kl. 11.20 skrev Andres Gomez:
Hello List
I am very interested in developing a research project on security
protocol for VoIP, under the GPL.
For some time I have been reviewing ZRTP, I would like to know the
Johansson Olle E wrote:
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
I realize that 1.4 at release time wasn't ready for
15 dec 2007 kl. 14.48 skrev Olivier:
Hi Olle
2007/12/15, Olle E Johansson [EMAIL PROTECTED]:
14 dec 2007 kl. 11.20 skrev Andres Gomez:
Hello List
I am very interested in developing a research project on security
protocol for VoIP, under the GPL.
For some time I have been
15 dec 2007 kl. 15.42 skrev Steve Totaro:
Johansson Olle E wrote:
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between
1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new
functions.
I realize
One of the biggest barriers to upgrading are the number of little
gotchas in syntax changes that can make an upgrade from 1.2 to 1.4
quite painful. After the pain I went through upgrading to 1.4, I've
always been recommending to people to think twice about upgrading if 1.2
does what they require.
Windows is a half-baked, dying OS that in essence is
a 32 bit extension and graphical shell, for a 16 bit
patch to an 8 bit operating system, originally coded
for a 4 bit microprocessor, written by a 2 bit
company, that can't stand 1 bit of competition.
Line of the year
On Saturday 15 December 2007 10:02:23 Rob Hillis wrote:
One of the biggest barriers to upgrading are the number of little
gotchas in syntax changes that can make an upgrade from 1.2 to 1.4
quite painful. After the pain I went through upgrading to 1.4, I've
always been recommending to people
Dovid B wrote:
Windows is a half-baked, dying OS that in essence is
a 32 bit extension and graphical shell, for a 16 bit
patch to an 8 bit operating system, originally coded
for a 4 bit microprocessor, written by a 2 bit
company, that can't stand 1 bit of competition.
Line of the year
On Sat, 15 Dec 2007 08:30:09 +0100, randulo wrote:
It's funny, but though I think nothing of having a linux box as a pbx,
on 24/7 for years, I can't imagine using windows this way. I think
there's little or no market for this whereas if there were a fanless,
diskless embedded solution for just
http://www.h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf
- Original Message -
From: satish patel
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 12, 2007 2:09 PM
Subject: [asterisk-users] Call Center Setup on asterisk
Dear all
I need call
Tilghman Lesher wrote:
If anything broke from the transition from 1.2 to 1.4, it is because you were
using something that was deprecated in 1.2. What we had attempted to do
in deprecation modes was to print the warning ONCE for each deprecated
operation, per Asterisk startup. I think that
The DNS for www.voip-info.org seems to be non-responsive. Is there a
mirror of this invaluable resource site?
Tx,
Steve
dig www.voip-info.org
;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server
; DiG 9.4.1-P1 www.voip-info.org
;; global options: printcmd
;; Got answer:
;;
I wonder if there are any major obstacles for upgrading.
From our perspective I'd have to say package management.
We manage a *lot* of asterisk boxes at client locations at the end of DSL
connections. We have a schedule to make sure each box is updated once a month
(e.g. these 10 boxes are
Steve Johnson wrote:
The DNS for www.voip-info.org seems to be non-responsive. Is there a
mirror of this invaluable resource site?
There are several mirrors, see
http://www.google.com/search?q=cache:www.voip-info.org/wiki/index.php%3Fpage%3DVoip-Info%2BMirrors
Regards,
Philipp Kempgen
--
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote:
On Saturday 15 December 2007 10:02:23 Rob Hillis wrote:
One of the biggest barriers to upgrading are the number of little
gotchas in syntax changes that can make an upgrade from 1.2 to 1.4
quite painful. After the pain I went
All I can say is with 1.6, if a change is made that causes something
that worked in 1.4 not to work in 1.6, please think twice, three
times or four times before making the change, or making the change
in such a way that it won't break dialplan stuff from 1.4.
Our policy is to never
On Sat, Dec 15, 2007 at 06:11:47PM -, Chris Bagnall wrote:
I wonder if there are any major obstacles for upgrading.
From our perspective I'd have to say package management.
We manage a *lot* of asterisk boxes at client locations at the end of DSL
connections. We have a schedule to
At 10:14 AM 12/15/2007, you wrote:
So Digium, (I address the company since Tilghman now works for you) do
you have any plans to query the user community and determine what a
typical end user of the product needs? With the knowledge and skill that
exists in your organization it would seem trivial
DNS for www.Voip-info.org should be back online shortly.
In the meantime here are mirrors:
a.. SimpleVoip.info - Location: California, USA, Bandwidth: 100M, Updated
Nightly
b.. Malico Inc. - Location: Tao-Yuang, Taiwan, Bandwidth: 1Mbps, Updated
Daily
c.. Totalip - Location: Oslo,
Hello All ,
On Sat, 15 Dec 2007, Johansson Olle E wrote:
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
I realize that 1.4
Dear Kiven;
Actually it is default and not degault. Also, I was
doing the compilation remotely via the Putty. Another
thing, I did another senario and got another thing, as
below:
I copied /usr/local/lib to /usr/lib and then I
restarted asterisk, but when I come back to run it,
then it was
It seems that all the warnings about deprecated functions in 1.2 did
not give the desired effect - that users move from the 1.0 commands to
the new applications and functions in 1.2. That caused real problems
when going from 1.2 to 1.4, since the dialplans where still on 1.0
level, not 1.2
When Digium starts using 1.4 in ABE then I would consider using it
in a
production environment. All I ever hear is soon, and I have heard
that for months if not the whole year. Until Digium itself is
comfortable selling and supporting this version, then neither am I.
Steve,
That's very
If anything broke from the transition from 1.2 to 1.4, it is because you
were
using something that was deprecated in 1.2. What we had attempted to do
in deprecation modes was to print the warning ONCE for each deprecated
operation, per Asterisk startup. I think that this was much too
Tilghman Lesher wrote:
If anything broke from the transition from 1.2 to 1.4, it is because you were
using something that was deprecated in 1.2.
After thinking about it for a while this is not true.
Well, it's true for the dialplan.
Changing CALLERIDNUM to CALLERID(num) is easy.
But i guess
Johansson Olle E wrote:
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
I realize that 1.4 at release time wasn't ready for release,
For this market,
people don't want anything complicated. I would imagine the software
equivalent of a run-of-the-mill answering machine.
Which has existed, in one form or another, for years. I was using a
voice enabled faxmodem a decade ago to answer my phone. The software
that came with it
Which has existed, in one form or another, for years. I was using a
voice enabled faxmodem a decade ago to answer my phone. The software
that came with it (don't remember the name, but WinFax also does/did
this) even allowed for a simple IVR, for mailbox selection and whatnot.
The only things
- Original Message -
From: Ira [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Saturday, December 15, 2007 2:50 PM
Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
At
On Saturday 15 December 2007 13:50:35 Ira wrote:
A perfect example is the new dial plan function array(),
it has nothing to do with arrays, doesn't accomplish anything useful
that couldn't have been done by allowing commas in set(), or calling
it setmany(), and means if real arrays ever get
Hi Everyone,
I am attempting to migrate my org to Polycom desktop phones. I need to
find a one touch park method.
I am using SIP 2.2.0.0047, and BootRom 4.0.0.0.423
I have found a few methods in previous posts:
Like this one posted by Anthony Rodgers
Hi all,
There's a myriad of options these days and I haven't been keeping up to date
with what's respectable any longer.
I essentially need a provider that will provide me with one DID to start and
let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on
my end and have full
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
Olle E Johansson wrote:
All I can say is with 1.6, if a change is made that causes something
that worked in 1.4 not to work in 1.6, please think twice, three
times or four times before making the change, or making the change
in such a way that it won't break dialplan stuff from 1.4.
You've hit the nail on the head with the crux of the pain I went
through. Finding stuff that was broke that I didn't realise was broke
until someone bothered to tell me about it. I'm sure everyone is
familiar with just how often users report problems caused by themselves,
but don't report stuff
A
set of scripts were recently discovered in the trixbox line of PBX
products, which connect to a remote host every 24 hours, to retrieve an
arbitrary
list of commands to be executed locally. These scripts were added
under the guise of submitting 'anonymous usage statistics', however,
with the
Hi,
Thanks very much for your response.
I'm don't think setting reinvites on will fix your problem. The
only thing I can think of is that you use some sort of call parking
to park the call on SiteB's asterisk server and then have the person
at siteB pick up the call from the parking lot
I
My biggest gripe is that everything loaded and seemed to work. A
day later we found this did not work and discovered a syntax
change. A day later something else did not work, an other syntax
change. Why isn't there some pre-processor to check the syntax of
the config files? Would
Hi,
I've recently come across LDAPget (version 2.0rc1) and I've been trying to
get it functional in my test environment (Asterisk 1.4.15 and MS Active
Directory 2003) but I can't seem to get it working.
I put together a test extension to try to change the CALLERID(name) by way
of a LDAP query to
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