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...ne 12-16 (starting 10 AM Monday, ending noon friday) Options: dCAP exam friday afternoon, June 16th Price: 2.500 Euro (ex VAT). 200 Euro (ex VAT) for dCAP. All trainings are pre-paid. Register by e-mail to info@edvina.net today. For more information, please visit our web site. ** The Asterisk SIP Masterclass :: Building SIP infrastructures with Asterisk The Asterisk SIP Masterclass is a new class we're launching in July. It requires knowledge of Asterisk and starts on a higher level than the bootcamp. The class is held by * Olle E. Johansson, Asterisk SIP developer * Terry Wilson, a consult...

Friends, Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When travelling around like this, we often invite the community to come and meet us in a nice restaurant. We offer good company and fun discussions about Kamailio, SIP-router.org and Asterisk - but the drinks and food are on you. At least yours :-) Berlin is the city where Si...

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!

...ready. The next release is not that far away, so it's not a big thing. We won't wait over 1 year like we did between 1.0 and 1.2. This weekend, I'm leaving for my Training in New York. Next training is in Stockholm, Sweden in June, after that we're launching the Asterisk SIP Masterclass in Chicago in July - with 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 tes...

...invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julv?rt) and decorating the Christmas tree. Of course, you understand that there's an Asterisk asterisk on top of all those trees, right? :-) After Christmas, I'm running the new Asterisk SIP Masterclass together with Daniel Mierla here in Stockholm. He's one of the core OpenSER developers and it's going to be a great class. I'm sure we will locate a set of new interesting bugs in svn trunk during that week. I'm really looking forward to that training. (Hint: We still have a few ope...

Hi, on ISDN there are the numbering plans that indicate if it's an national or an internation number. Is there something similar on SIP? How should i set a callerid to an internation number? complete e164, with, without an intl prefix (ie +, 011, 00 etc)...? How to a national number? Regards, Andreas. _________________________________________________________________ Discover fun and

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Are Zphone and ZRTP the future for the Voip Security? Opinions?

hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can?t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone

So I have two classes for openvz setup. The first is openvz::setup and it sets up the box for OpenVZ.. then another class (in same .pp file) called openvz::master sets up the config for the main system (as opposed to a VE). The error I see is that the openvz::master class is executed (and fails) becuase it doesn''t seem to load up the setup class? debug: Calling puppetmaster.getconfig

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username