similar to: Development news :: Smarter medialess calls!

** Asterisk Bootcamp in Stockholm, Sweden The next Asterisk Training is the Edvina.net Asterisk Bootcamp - the class we have been giving for over a year under the brand name "Astricon Training". The same teacher, the same material and a new name. All students have a PC and will install a fully working Asterisk PBX. During the week, we will build a business PBX configuration as

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

* 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

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 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

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,

Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 *

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!

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

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))

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

Friends in the Asterisk community, Yesterday the Asterisk development branch, also known as "svn trunk", changed quite a lot. We added two major features: A new version of AEL and a new configuration system. Hang on, and I'll explain! * AEL - The Asterisk Extension Language -------------------------------------------------------- Last summer, Mark Spencer created a new

** Want to see who you're talking to? Video telephony is growing. A couple of developers has formed the Asterisk Video Task Force in order to improve the support for video telephony in Asterisk for the 1.6 release this fall. There is already support for video in the SIP and the IAX2 channel, but we need to add more in order to improve the support, among other things add video to H.323

I had this same problem with WINXP WinMESS, (what a name mess) I changed the Distro from Redhat 8.0 to Mandrake 9.1 and bam! It all works!! Does anyone know of a problem with this and RH 8.0???? Are you running Redhat?? I now have Messenger working fine as well as X-ten, Sipps, and some others. I have standardized on Mandrake 9.1 and asterisk seams to have NO problems. REDHat 8.0 proved as

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

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

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

Today is Swedens national day - since a few years a holiday too. We don't have a tradition on how to celebrate. Sweden has not been to war for a very long time, so there's no real spirit for the country here - it's been aroundfor such a long time, so what? :-) Guess we have to learn from abroad, to get a celebration feeling like July 4th in the US or May 17th in Norway (from

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

Spring is back in the Stockholm area. After a few day's worth winter re-runs, the sun is back and night-time temperature is at least 5 degrees celsius. Time to move out all my annual flowers and prepare the garden for summer. Sweden is famous for our annual five week holidays - by law. From june to late august, it's almost impossible to make any business decision, since there's always