similar to: SIP channel modification

** 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

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))

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

*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.

How can a reload clear registrations? If I 'reload' without using realtime, I keep my sip peers (as well as astdb). I can still contact other phones... registration info is still there. If I `reload` with realtime, I lose my sip peers (but astdb remains). I can _STILL_ contact other phones.... registration info is still there and Asterisk must be referring to astdb to find the IP

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

---------------------------------------------------------------------- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <oej@edvina.net> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - <asterisk-users@lists.digium.com> Message-ID:

--- "Olle E. Johansson" <oej@edvina.net> wrote: > From: "Olle E. Johansson" <oej@edvina.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk on FreeBSD > Date: Mon, 27 Oct 2003 08:24:22 +0100 > > Rich Adamson wrote: > > >>My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD > server.

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

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

Hello Thanks for the answer... Just realized it myself, as your mail arrived :) Could be a nice feature though. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using

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.

* 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.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

Friends, For a long time, we needed a platform for managing feature requests - things that the community or developers would like to see in Asterisk. In the bug tracker we used to have a "feature request" category, but there was no good way to handle them in the bug tracker and they where in the way for the work done by developers in the tracker. The new site is basically a blog

** 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

Friends, After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code. After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working

Yes, the thanksgiving holiday is here (in the USA)! But also, the fear of running out of IP addresses next year has raised its ugly head and since we don't do Thanksgiving in Europe, we have some serious talking to do about this problem. This Friday at 12 Noon EST, Olle Johansson will be joining us to describe the state of the migration to IP v6 in VoIP-dom. Olle (@oej) needs no introduction.