similar to: Wiki - stable -dev

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.

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))

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

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 =>

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

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 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

* 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.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

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,

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

** 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

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

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!

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have

Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

*lol* I think he was referring to a ASTERISK reboot, not a phone reboot. > -----Original Message----- > From: Olle E Johansson [mailto:oej@edvina.net] > Sent: Thursday, March 23, 2006 1:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Subscription state after reload (New > subject) > > > > 23 mar 2006 kl.