similar to: Asterisk Users in Madrid?

We're getting close to Astricon Europe 2005, the first Asterisk Community gathering in Europe. Speakers are coming in from all over the US and Europe, as well as far away as New Zealand, to talk, teach and discuss Asterisk -the Open Source PBX. At this time, we're still accepting registrations online, where you reserve a hotel room and pay for the conference. Make sure you register now,

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

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one

-- GOOD NEWS! -- As ground transportation and the costs associated therewith are something of an issue. So, in the spirit of making AstriCon available I've contacted the hotel and arranged/discovered some low cost options: -- MARTA -- MARTA, the Atlanta mass-transit system (i.e. train) can get you there. Take MARTA from the airport north to the Brookhaven station. The hotel shuttle will

> > Message: 2 > Subject: RE: [Asterisk-Users] Astricon costs... > From: Steven Critchfield <critch@basesys.com> > To: asterisk-users@lists.digium.com > Date: Wed, 21 Jul 2004 21:30:26 -0500 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-07-21 at 19:34, Steven Sokol wrote: > > > Has anyone really looked at the costs for Astricon. But the

Has anyone really looked at the costs for Astricon. But the hotel costs. $111.00 USD per night.. come on guys give me a break. I will not be staying at that hotel. I can rent a car and stay near the air port for almost half that. In addition from what I have been told their will be no shuttle service from the Airport to the hotel. Anyone else have any input on this? bkw_ PS: I'm going

For those of you coming from out of the area and who may not be familiar with the bay area. TRAVEL: The closest airport is San Jose International Airport (SJC; 10 miles away). There is also San Francisco Internation Airport (SFO; 43 miles away) or Oakland (OAK; 49 miles away). IMHO, I would choose SJC, then SFO, and lastly OAK, depending upont he prices. HOTELS: The Cypress Hotel

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 Bill, On Fri, 2007-03-23 at 15:57 -0700, Bill Wendling wrote: > For those of you coming from out of the area and who may not be > familiar with the bay area. > > TRAVEL: > > The closest airport is San Jose International Airport (SJC; 10 miles > away). There is also San Francisco Internation Airport (SFO; 43 miles > away) or Oakland (OAK; 49 miles away). IMHO, I would

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,

** 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

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!