search for: oll

...sion 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 the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. /Olle

...ere is code for testing available. 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/

...ou consider the most important new feature 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/

Rich and I have updated the Wiki page "Asterisk rollout tips" with advice on how to plan and implement your Asterisk rollout. This page is based on many discussions on the mailing list, so don't be surprised if your comment or thought is included in the text. Thank you for your input! http://www.voip-info.org/tiki-index.php?page=Asterisk+ro...

...ettings to indications.conf and started working on a number of i18n efforts for Asterisk. A big thank you to all of you that helped me with this, and to all of you that contributed code! Finally, I can confuse my Swedish customers by randomly telling numbers in Italian, French or Portuguese :-) /Olle

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

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

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) || (re...

...w, downloading it is a good way of wasting the bytes on your hard disk drive and not much more. In Q1 2007 I will run an AstriSIPcon developer's meeting to be able to meet everyone that has interest in Asterisk and SIP to test, discuss and 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...

...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 chan_sip2.so: chan_sip2.o cd /usr/src/asterisk make make install I assume that problem is with what did or didn't add to the Makefile Thank for any help ----- Original Message ----- From: "Olle E. Johansson" <oej@edvina.net> To: "...

...? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara (NuFone) and collegues from the SIP Foundry Open Source project. Main topics: * Integrating the PBX with the IT infrastructure: Asterisk for the Enterprise * VOIP migration in-a-box: Asterisk for Service providers * Lower cost, more flexibility: Asterisk for Call Centers * Your VoIP Swiss Army Knife: Asterisk for...

...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 and look at the code, but I know f^ck all about C programming, I'm a Shell man myself! -----Original Message----- From: Olle E. Johansson [mailto:oej@edvina.net] Sent: Thursday, 8 January 2004 6:42 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP reload configuration problem /* New subject */ Christopher Raper wrote: > I have also noticed that sip.conf doesnt get updated without a restart......

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're taking are: * First a quick-fix only for saying numbers * Adding documentation and sample sound files Many patches require additional sound files compared with the engl...

...33i 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 are under the impression that BLF DOES work > after a reload or reboot provided the phone re-registers? > > I have two separate manufacturers phones. Aastra 9133i and Grandstream > GXP2000 and both behave EXACTLY the same way. After a reloa...

...siconhold, agi/fastagi... New interesting challenges. So take one of these standard rack servers from HP and run a telco for a small city on one box. While you're at it, buy a spare one, hardware can fail ( ;-) ). But don't say that Asterisk does not scale well. Those times are gone. /Olle --- * Olle E Johansson - oej at edvina.net * Open Unified Communication - SIP & XMPP projects

...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 little dollar is worth quite a lot! I fully trust that Joachim (Zoa) and his team will complete this in a good way and look forward to improved sound quality in the SIP channel. Read more here: http://www.astertest.com/forum/viewtopic.php?t=13 Thank you for your contribution! /Olle If you're going to...

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 =>

...ith a gold team teaching: Ed Guy, Terry Wilson and myself. While I'm travelling around, you can spend all your free time testing Asterisk 1.4 for us. We need your help, now. Download svn trunk and test in your environment! On behalf of the community - thank you for testing! SIP greetings! /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/

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

...e useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a try. Now, to be the documentation-pain-in-the-*** I would like to get an explanation of the autocreatepeer SIP.conf setting and functionality? It's not in sip.conf.sample yet. /Olle