similar to: SIPit 26 in Sweden - organized by Edvina

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

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

We have here a TandBerg "videoconferencing system" connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtaining audio calls, the video not appears. When i try to make a 64K calls (data calls), the call does not

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this -

Hi, For some time now (starting with the 3.10 or 3.11 kernels, but I'm not sure) I have the following problem: I have a machine with 6 SATA slots and two SAS controllers, one onboard HBA and one RAID controller in a PCIe slot. The problem is that the order of the SAS controllers changes randomly after a reboot, so the SCSI addresses of the devices change. One time lsscsi prints (empty

** 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

* 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

*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

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

Want to become an Asterisk SIPmaster? Register for the Asterisk SIP Master Class, taking place in Chicago, IL, USA July 10-14 organized by Edvina in partnership with Digium. We're developing this new training now, creating labs with Asterisk and SIP express router, NAT traversals, realtime and much, much more. Learn more here: http://edvina.net/training/sipmasterclass/ and register

hi, If i try a backup with tar / tape sdlt320 on a debian-sarge / 2.6.12 based XEN 3 System, the System freeze. If i test this by knoppix the backup works fine. Test with xen 2.07, Kernel 2.6.11.12 on xen 3.0.0, Kernel 2.6.12 This System in a Interserver with an LSI scsi-Controller and a Tandberg SDLT320 Streamer. Are thereany known issues with this config and xen ? regards Rene

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

---------------------------------------------------------------------- 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:

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

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))

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

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