search for: edvina

...usiness would not be as large as it is today. Without working and tested standards, it would not work at all. Asterisk, as an Open Source platform is in the middle of this business. We simply have to interoperate with all kinds of phones, servers and services out there. * SIPit 26 - organized by Edvina SIPit is organized by the SIP Forum and every SIPit - two per year - is hosted by a company. During good times, the large vendors has taken care of this. In the current climate, it's hard to get the needed resources - time and money - from these vendors. SIPit 25, yes the 25th in a successful...

* 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 add audio and video capabilities to microblogging,...

** 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 co...

...d on this from someone using Polycom phones so it is > happening on at least 3 totally different phones. > <http://bugs.digium.com/view.php?id=6047> > > Sorry if I am hijacking this thread. > > > -----Original Message----- > > From: Olle E Johansson [mailto:oej at edvina.net <http://lists.digium.com/mailman/listinfo/asterisk-users>] > > Sent: Thursday, March 23, 2006 12:15 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Re: Subscription state after > > reload (New subject) > &gt...

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 =>

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

...004 1:32 PM > To: asterisk-biz@lists.digium.com > Subject: [Asterisk-biz] Asterisk training and certification :: > AstriconTraining > > *** AsteriskT Open Source Linux PBX Training and Certification > > Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc., > Edvina AB and Sokol & Associates today released a new program for > training and certification of Asterisk professionals. Asterisk is the > leading Open Source PBX for Linux, with support for both PSTN > connectivity and many VoIP protocols. > > The first class in the Astricon Training...

...0) ms = 0; return ms; gettimeofday might be failing!! Why? I'd add a call to perror() after "his should never happen" I'll bet it has to do with time zones But then maybe ms is beingset to zero... Good luck --- "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 21:25:04 +0100 > > Perry E. Metzger wrote: > > > "Olle E. Johansson"...

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!

...ature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - oej at edvina.net * Asterisk Training http://edvina.net/training/

...nd work with the new SIP channel. SIP greetings! /Olle PS. A big thank you to Voop AS, who keeps supporting my development work with Asterisk as well as all the students in my training classes that provide development funding by attending the classes. Thanks! --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/ * Next class: Stockholm, Sweden November 13-17 2006

*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. 20.09 skrev mustardman29: > > > Sorry for the...

...eload` 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 address the other phone is at. Doug > -----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. 20.09 skrev mustardman29: > > > Sorry for the...

...- Totally rewritten SIP transfer code In testing right now is the new t38 passthrough code, which will be integrated after some more reviews and tests. Please help us test that code too. Thank you all for working hard on this until I return online! Regards, /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/ * Next training and dCAP in Stockholm, Sweden, June 2006! --- * Olle E Johansson - oej@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden

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 today! Questions? E-mail info@edvina.net today! I...

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

...If you are interested, please check this URL in the bug tracker: http://bugs.digium.com/view.php?id=5090 I think this is a big step for Asterisk. Do you? If so, don't forget to say "thank you" to Steve Underwood - Coppice! Have a nice weekend! /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/

---------------------------------------------------------------------- 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: <A8C949D0-6208-41FF-85BD-E8BDDA6BFCF5@edvina.net> Content-Type: text/plain; charset=US-A...

...gium added the voice of the southern gentleman Danny Wyndham and the Swenglish dialect of Asterisk developer and guru Olle E. Johansson, one that was recognized with a strange smile by all Asterisk developers testing VCC. VCCnet technology includes scalability and security components licensed by Edvina AB in Sweden. Edvina's experience of large scale Unified Communication networks was necessary to build a world-wide network-centric platform for this new service. - "We find it exciting to contribute to this new service. Realizing the perfect match between the open IPv6 protocol and the...