similar to: Discussion: Are we ready to leave 1.4 behind?

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

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

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.

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

Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get

Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is

Rather hasty I think. I think whatever version 1.2.X winds up on should be the most stable release of Asterisk, period. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Olle E Johansson > Sent: Tuesday, May 29, 2007 5:19 AM >

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

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,

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

FOR IMMEDIATE RELEASE Puerto Escondido, Mexico, April 1st, 2010: Digium launches Asterisk VCC (TM) - a new virtual communication platform for enterprises, the public sector and the home. =========================================================== Asterisk 1.8 will contain a stunning new technology for all Asterisk users world- wide - virtual communication clouds or VCC (TM). With this

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ /O

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

* 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

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

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

Friends, At the first Astricon I was very happy to see Marc Blanchet as one of the attendees. I knew he was one of the IPv6 gurus and wanted someone to show some interest in Asterisk and IPv6. Well, he did not only get interested in it, but started coding on it. The results have been available for quite some time at http://www.asteriskv6.org/ and Marc has tested it at several SIPits for

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