Displaying 20 results from an estimated 10000 matches similar to: "*** Hang on, we're on our way to 1.0"
2006 Apr 05
0
The Asterisk bug tracker :: please think twice before opening a report!
Friends,
At this point, we're close to 300 issues open in the bug tracker
at http://bugs.digium.com
Some of us spend many hours each week,
if not each day, to work with the bug tracker. It's a tool for us, a
very important tool to handle new features
and find bugs in Asterisk, tracking them down.
It is important that you consider a few things while using this tool:
- If a bug marshal
2006 Oct 11
4
Psst... Top secret information: Codename Pineapple
Friends in the Asterisk community,
I've been talking for years about the new version of the SIP channel.
I've been trying to get funding
and get going. Well, the funding part remains to be handled, but I
have other news - if you kan keep
it to yourself.
...I've began coding. Finally.
With a happy smile on my face I removed "pedantic=yes" the other day.
After years of
2003 Oct 27
1
Fwd: Re: Asterisk on FreeBSD
Your log file almost looks like a bug in Asterisk doesn't it?
Why call poll() with a zero timeout while passing only one FD?
and then why do the read when there is no data?
Read the man pages for all the system calls
Take a look at the source chan_sip.c
/* Wait for sched or io */
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
2006 Mar 24
1
Re: Subscription state after reload (New subject)
Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own.
> Thanks Olle,
>
> So am I to understand that you
2004 Apr 20
1
Re: SIP re-invite
Trouble getting chan_sip2 to compile
below is what I have done
-download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software
cp /root/software/chan_sip2s.c /usr/src/asterisk/channels
cd /usr/src/asterisk/
patch -p0 acl.c /root/software/acl.c.patch
cd /usr/src/asterisk/include/asterisk
patch -p0 acl.h /root/software/acl.h.patch
- added the follow to /usr/src/asterisk/channels/Makefile
2006 Mar 10
3
Development news :: T38 passthrough support
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
your area or give you more
consulting oppurtunities - in short, functionality that will make a
lot of sense for you users.
However, developers can't really get anywhere without a
2005 Jul 23
3
Asterisk 1.2 is getting closer - please help
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
been almost a year and we haven't been able to go ahead and release a
new version. Now is the time to try to move forward again.
As we've outlined before, the process is this:
--------------------------------------------------------------------
* Code freeze: At this point, we'll
2006 Jan 25
14
No audio? Update your Asterisk
This morning we discovered a serious bug that stopped all bridged audio
in our Asterisk servers. Mark found the problem and soon fixed it.
If you get this problem today, please update your Asterisk server. A fix
has been commited to the subversion repository for 1.2 as well as trunk.
A fixed 1.2.3 release will be published on ftp.digium.com as soon as we
can find a release engineer (consider
2007 Dec 15
17
Upgrade to Asterisk 1.4 - it's one year's old!
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 polishing it,
working hard with bug fixes. The 1.4 that is in distribution now is
2009 Aug 25
6
Breaking news, but what happened? 11.000 channels on one server
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny boxes and other
interesting solutions. Testing has been a large part of these
projects. How much can we put into one Asterisk box? Calls per euro
invested matters.
So far,
2004 Jun 13
4
*** Asterisk Sunday News: Off track with 1.0, moving forward
Thank you very much for all feedback on Asterisk Sunday News!
This is the last issue for June. This week I'll go on holiday
and will be back with more news in early July.
My kids are getting summer leave this week and we'll be
visiting the south of England for a while. Another part of
Europe that still use their own currency.
If you think there's an European standard, you're
2006 May 19
1
Development news :: Smarter medialess calls!
Friends,
To update you on recent changes in svn trunk, I can inform you that
we now have ever smarter
ways to handle media streams in Asterisk than we do in 1.2 for the
IAX2 and SIP protocols.
* IAX2: Splitting signalling and media apart
Starting with the IAX2 protocol, we now have the ability to transfer
media streams to go directly
between IAX2 servers and keep the signalling path.
2009 Apr 01
10
FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
In a surprising move, Digium in partnership with Edvina today released
a new channel driver for Asterisk, chan_tweet. The driver connects
seamlessly to several microblogging platforms, including Twitter,
Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of
this new module is to
2004 Jan 08
2
SIP reload configuration problem /* New subject */
When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear.
As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though!
Check it out and let me know what you get.
Cheers
Chris
PS - I would try
2009 Sep 03
3
GTalk functionality Asterisk
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like connection
break etc... so i left them ......... and start wondering if there is some
thing that dont require configuration at router layer. The task to
accomplish to make and recieve calls from outside local network using any
protocol whose soft phones are
2006 Apr 05
5
Dial Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten =>
2008 Jan 12
1
Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!
I've written a new article about Asterisk 1.4's Jabber integration.
Check it out at
http://www.voip-forum.com/asterisk/2008-01/xmpp/
/Olle
2004 Apr 28
5
Asterisk goes international :-)
During the recent week, we've worked hard to add more of the contributed international
support to Asterisk. A big step was taken yesterday when Mark added international support
for saynumber() to CVS. We now have a first version of support for
* Danish
* German
* English
* Swedish
* Norwegian
* Portuguese
* Italian
* French
All of these require that you add your own sound files. There are
2004 Apr 19
3
One, två, tre, quatre, cinq ... International numbers in say.c
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
* Support for other language syntaxes in saynumber
Accidentally I opened this can of worms to see if we can add support
for other language syntaxes for saying numbers. Seems like Swedish,
english and norwegian follow the same syntax. I've integrated
existing patches for french, danish and soon portuguese syntax.
The steps we're
2008 Feb 13
3
What is a "secure call"?
Friends,
The following mail was sent earlier to asterisk-dev and did not cause
the amount of discussion I hoped it would.
Now that we have a way to secure signalling in IAX2 and SIP in
Asterisk svn trunk, we need to start working on
the concept of a "secure call" - or does it really matter?
In SIP, there's a specification for how I as a domain owner can
request all calls to