[asterisk-biz] Do you develop SIP software or work heavily with your own Asterisk-based platform?
If so, please consider attending SIPit - the SIP interoperability test event held this September in the UNH Interoperability labs in Durham, New Hampshire, USA. At SIPit we test SIP implementations - it’s not a certification, but peer-to-peer tests and multiparty tests. We have a great testbed of various NAT networks for those of you working with NAT traversal issues, we have IPv6 in our network as well as an extensive test of TLS. This year we will also focus on STIR - the new secure identity handling in SIP. During one week, we’ll work in a non-competitive environment to enhance implementations and improve standards. Read more about why you should participate here: http://www.slideshare.net/oej/participate-in-sipit To register and learn more details got to the main SIPit web site https://www.sipit.net/Main_Page SIPit is organised by the SIP forum If you have any questions, please don’t hesitate to ask! /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Want to learn Kamailio from the ground up? The Edvina SIP Masterclass in Madrid Sept 15-19 is open for registration!
Hi! Kamailio is the leading Open Source SIP server. You can use it as a proxy, a presence server, registrar and much more. It's a toolbox for creating SIP server applications. Scalability and performance is at the core of this SIP server. But it is very different from Asterisk or FreeSwitch - or any SIP PBX you have been running before. To control it, you process SIP messages in Kamailio's own programming language. This means, you have to understand the SIP protocol. My SIP Masterclass is the quick start for you. Five days of lessons and labs covering the SIP protocol, RTP media, NAT traversal and how to get Kamailio to work in your network. Next class is in Madrid and it's priced at a summer special price - 2.500 Euro ex VAT for five days of SIP and Kamailio. Read more about the class on our web site. If you have any questions or want to book a seat, just give me a call or send me an e-mail. We already have many registrations, so this class is ready to go. Students getting ready to learn how to build secure, scalable and resiliant SIP platforms. It's time for you to attend! http://edvina.net/training/new-sip-masterclass/ I am also running this class - and many other related classes - in-house for companies that need a combined workshop and training session. If you are interested in such an event, just contact me and we'll set it up. Looking forward to seeing you in Madrid! Best regards, /Olle E. Johansson --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Time to scale your Asterisk with Kamailio? Attend my training this summer!
Hi! The reason why I am writing to you is that I am preparing for the next SIP Masterclass and still have a few seats available for you or your team members. The class is administered by Avanzada 7 and held in beautiful Torremolinos, just outside Malaga, Spain. We start June 30 and end after five days of lessons, labs and SIP fun on Friday July 4th. This class is for the Kamailio beginner, someone that has been running Asterisk for some time and need to figure out how to scale, how to add new services and how to configure and run Kamailio. You will learn how and why you use Kamailio in your network. How you install, configure and run it. How you handle NAT traversal and how you integrate it with Asterisk or FreeSwitch. Students run practical labs in virtual machines on their own laptops, being able to take all the configurations back home and get them running in a real-life SIP network. Every day we mix labs with lessons, making practical use of the knowledge gathered. The class begins with the SIP protocol from a basic level, something that you must understand in order to configure and run Kamailio. From there we work ourselves up the stack, covering RTP, RTCP, QoS, NAT traversal and of course security. The class builds on my 10 years of VoIP training experience that started with the Asterisk Bootcamps and then the Asterisk SIP Masterclasses. It's my third generation of training class - one that focuses on Kamailio and SIP. There will be a lot of Asterisk in there too, I'm still working with Asterisk on a daily basis as a developer. If you have attended one of my Asterisk bootcamps or Asterisk MasterClasses before, this is a new class, an upgrade. You can learn more about the class at http://edvina.net/training/new-sip-masterclass/ If you have any questions or just want to register - please contact me directly! If this class doesn't fit the needs of your team, I quite often run in-house trainings on site. That way, I can select the training material and pace that fits your group. I have training material on a wide range of topics, from TCP/IP basics to SIP, Asterisk, Kamailio and WebRTC - including development with Asterisk and Kamailio. It's a great way to bootstrap a new project or raise the shared knowledge in a group. Don't hesitate to contact me if this would work better. Looking forward to seeing you in Malaga! /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Learn how to build scalable SIP platforms - SIP Masterclass in Miami, FL this December!
Hi! The Edvina SIP Masterclass is a one-week training class in SIP and Kamailio. How to build scalable SIP platforms, how to traverse NAT, how to handle failover and load balancing and much more. It's the perfect training for companies who have built Asterisk platforms and needs to learn how to scale better. This class is for users of Asterisk and FreeSwitch that wants to learn how to build larger, scalable and open SIP networks with Kamailio – the Open Source SIP server. The class interactively teaches you SIP and Kamailio, building a platform step by step. When you leave the class, you should know much more about how SIP works and how Kamailio can scale your existing solution or be the new platform for a Unified Communication network. • The SIP Protocol • Kamailio – the SIP server • SIP call flows: Call transfers • SIP: Forking and routing • Kamailio – transactions and forking • SIP Media: RTP, RTCP and QoS issues • SIP NAT traversal: Stun, Turn, Outbound • SIP presence infrastructure: SUBSCRIBE, NOTIFY, PUBLISH • SIP Dialogs, dialog states, blinking lamps • SIP messaging and presence: SIMPLE and MSRP • Kamailio messaging and presence • Building SIP services with Kamailio and a media server (Asterisk, FreeSwitch) • SIP load balancing and failover, DNS • Kamailio: DNS, failover with Dispatcher • SIP security: TLS, S/MIME, SRTP, SIP identity It's the first time for many years in the USA - so register now for this unique oppurtunity! http://edvina.net/blog/2013/09/learn-sip-and-kamailio-in-miami-this-december/ See you in Miami! /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Edvina and Avanzada 7 offers grant for participation in the SIP Masterclass, July 2013, Malaga, Spain
The next Edvina SIP Masterclass featuring Kamailio and SIP will take place the first week of July in Malaga, Spain. One week of labs, lessons and interaction with other people working with realtime communication will be a boost to your career, and give you insights into how you build scalable infrastructure platforms for SIP communication. It’s a lot of fun and a great learning experience for everyone. But not everyone that needs it can afford participating, which is why we are offering a SIP Masterclass Grant programme this year. Since there are hard times with a high rate of unemployment in Southern Europe, Edvina and Avanzada 7 has decided to open up a grant for a free seat in the class. To apply for this, you have to be unemployed and have previous experience with realtime communication – using Asterisk or FreeSwitch or maybe even OpenSER or Kamailio. Edvina and Avanzada 7 will make the decision about who gets the opportunity to participate for free. You will have to pay travel and your stay, we will give you a no-cost participation in this class. Read more at this url: http://edvina.net/blog/2013/05/sip-masterclass-grants/ In addition, we're open for registrations for the class. For payments, there are options to divide the fee into multiple installments. Check with Avanzada7 and find a solution that works for your company! http://edvina.net/blog/2013/01/sipmaster-malaga-2013/ See you in Malaga! /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Reverse bounty - funding needed for SIPADDHEADERRESPONSE()
Friends, Many people have asked me for the ability to add headers to a response to an INVITE. With a function, like SIPADDHEADERRESPONSE() you can add a header in the dialplan that will be copied to all 1xx class responses after that dialplan entry, as well as in the final response to the INVITE. In many cases this is for call tracking with Voip Monitor or Homer SIP Capture - but also for other purposes. In order to complete such a branch for 1.8, 11 and trunk I would need two days of funding for coding, managing the process of reviewboard and testing. If you are interested to contribute at least 150 USD or more please contact me off list. The fee would be billed from my company. Thanks for your support, /Olle --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] The Edvina SIP Masterclass v2.0 in Miami FL - Dec 10-14 2012
Friends, In December, I'm running the new Edvina SIP Masterclass in Miami Florida. This is the first time in many years I'm running a training in the USA. Want to learn more about the OpenSER/Kamailio SIP Server and the SIP Protocol through labs and presentations given by a teacher with ten years of experience in building large scale SIP networks? Register now to get seats! Email i...@edvina.net for details. * WHAT IS THIS CLASS? I've been running a class named "The Asterisk SIP Masterclass" for many years, as well as a large number of in-house classes for developers, VoIP management teams and call centers. When you do that, you add information to the slide deck every time. And you keep adding. After a few years, you have too many slides for the class and you gotta change. So that's exactly what I did. I've removed the Asterisk part and added more about SIP and much, much more about Kamailio and how to operate Kamailio in a network with a media server like FreeSwitch, Asterisk or a proprietary box. This class is for users of Asterisk and FreeSwitch that wants to learn how to build larger, scalable and open SIP networks with Kamailio – the Open Source SIP server. The class interactively teaches you SIP and Kamailio, building a platform step by step. When you leave the class, you should know much more about how SIP works and how Kamailio can scale your existing solution or be the new platform for a Unified Communication network. * WHAT IS THE CONTENT? • The SIP Protocol • Kamailio – the SIP server • SIP call flows: Call transfers • SIP: Forking and routing • Kamailio – transactions and forking • SIP Media: RTP, RTCP and QoS issues • SIP NAT traversal: Stun, Turn, Outbound • SIP presence infrastructure: SUBSCRIBE, NOTIFY, PUBLISH • SIP Dialogs, dialog states, blinking lamps • SIP messaging and presence: SIMPLE and MSRP • Kamailio messaging and presence • Building SIP services with Kamailio and a media server (Asterisk, FreeSwitch) • SIP load balancing and failover, DNS • Kamailio: DNS, failover with Dispatcher • SIP security: TLS, S/MIME, SRTP, SIP identity In the class, we will mix generic SIP presentations with implementation-specific presentations covering Kamailio and then put everything together in labs. You bring your own laptop with a Linux virtual machine. You are of course free to bring any SIP devices you have and want to use during the training! Contact me today if you have any questions or just want to reserve a seat. Details about location, prices and the class is available on the Edvina web site at this address: http://edvina.net/training/new-sip-masterclass/ Looking forward to meeting you in Florida! SIP-greetings! /Olle Johansson -- * Olle E. Johansson - o...@edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 * http://edvina.net/training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] The New SIP Masterclass from Edvina - Stockholm, Sweden Oct 15-19 - register today!
Friends, The New SIP Masterclass will be held in Stockholm, Sweden Oct 15-19. As this is the first run of this new and upgraded training in SIP and Kamailio, we're offering great discounts. 20% for all students and 30% for previous students in our classes. This class is built for persons that have used the PBX-class tools like Asterisk, Yate and FreeSwitch and wants to learn how to scale and add new applications like presense and instant messaging to their solution. The class will spend a lot of time on the SIP standards, then move on to how to implement them using Kamailio – the Open Source SIP server – in combination with other tools. After the class, you will not only know how to operate Kamailio – you will also have a lot of knowledge about how the SIP standard works, what to expect from devices and how to troubleshoot your realtime network. Read more about the class on our web page! http://edvina.net/training/new-sip-masterclass/ If you want to register, just send me an e-mail! Regards, /Olle smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Launching: The NEW Edvina SIP Masterclass in Stockholm and Miami
Friends, Yesterday Edvina, my company, launched the new and updated SIP Masterclass training. It is scheduled for Stockholm in October 2012 and Miami, Florida in December 2012 (thanks to Redfone Communication). The new SIP Masterclass is partly based on the old Asterisk SIP Masterclass, but focuses more on Kamailio - the leading Open Source SIP server - and the SIP protocol. We've added a lot of Kamailio information and new labs, as well as information about SIP updates like Outbound, GRUU, ICE, Turn and much more. * A HIGH LEVEL TRAINING IN SIP AND BUILDING SCALABLE REALTIME PLATFORMS The Edvina SIP Masterclass is aimed at people who work professionally with Asterisk, FreeSwitch or other systems and wants to learn more about how to operate a SIP proxy - Kamailio - and get scalability, new services and failover for their platforms. The class also gives a thorough understanding of the SIP protocol, from the basics to advanced functionality like dialog states, SIP transfers and NAT traversal. • The SIP Protocol • Kamailio – the SIP server • SIP call flows: Call transfers • SIP: Forking and routing • Kamailio – transactions and forking • SIP Media: RTP, RTCP and QoS issues • SIP NAT traversal: Stun, Turn, Outbound • SIP presence infrastructure: SUBSCRIBE, NOTIFY, PUBLISH • SIP Dialogs, dialog states, blinking lamps • SIP messaging and presence: SIMPLE and MSRP • Kamailio messaging and presence • Building SIP services with Kamailio and a media server (Asterisk, FreeSwitch) • SIP load balancing and failover, DNS • Kamailio: DNS, failover with Dispatcher • SIP security: TLS, S/MIME, SRTP, SIP identity * A NEW CLASS BASED ON MANY YEARS OF EXPERIENCE The old class has been running for many years with good feedback from the students. The classes are based on years of experience on building scalable platforms for enterprises, the public sector, call centers, service providers and universities. The class starts on a level where basic Elastix, FreePBX, Asterisk and FreeSwitch trainings leave the student. Read more about the class here: - http://edvina.net/blog/2012/08/new-sip-masterclass/ - http://edvina.net/training/new-sip-masterclass/ * DISCOUNT ON THE STOCKHOLM CLASS! We have a *SPECIAL OFFER* for the first class in Stockholm, since this is the first class with the new material and the new labs. New students get a 20% discount and students that have attended Edvina trainings before get 30% discount! Contact us today via e-mail to reserve a seat! If you have any questions, please don't hesitate to contact me! Regards, /Olle smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Last minute bookings - Asterisk SIP Masterclass in Barcelona
Friends in the community, Thanks to some unfortunate last-minute cancellations we have a few seats available on the Asterisk SIP Masterclass in Barcelona next week. This class goes through SIP, Asterisk and Kamailio. It starts where the Asterisk classes and dCAP leaves you and introduces Kamailio on a basic level. We're focusing on how to build scalable SIP networks. * The SIP Protocol * ASterisk SIP channel * Kamailio/OpenSER/SIP-router * Debugging your installation * NAT traversal * Scaling Asterisk * Large scale platforms with Kamailio and Asterisk * SIP presence with Asterisk and Kamailio And much more. To spend one week with a gang of VoIP professionals will give you a lot of input, inspiration and new knowledge. This is the very LAST TIME I'm teaching Asterisk in a public class. After this I will only do Asterisk trainings in in-house trainings. (And today is a perfect day to negotiate pricing with me...) Feel free to contact me to register today! For more information, please read http://edvina.net/training/sipmasterclass/ See you in Barcelona! Regards, /Olle --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Digium's new Community Support Manager - Rusty Newton
25 maj 2012 kl. 16:41 skrev Kevin P. Fleming: > We'd like you all to help us welcome Rusty Newton to Digium's Asterisk > development and community support team! Rusty has been with Digium for > over five years, starting in the Technical Support department and then > moving to a sales position where he assisted customers with Asterisk and > Switchvox solutions to their business needs. Prior to joining Digium he > spent more than five years in the telecom industry, installing, > configuring and maintaining PBXs. A couple of weeks ago he moved into a > new role (for him and for Digium), Community Support Manager. > > In this role he'll be the primary person responsible for ensuring that > Digium's community services are providing what the community members > need, that the systems are operating properly, and that issues and > questions are getting the attention they deserve. He'll be working > closely with our Community Director as well, especially for events like > AstriCon and others. He works directly with the software development > team at Digium, which will allow him to focus almost exclusively on > technical issues and discussions. > > We're quite excited that he has taken on this role and we expect that > you will soon see the benefits of his activities across the community! Welcome Rusty! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Quote for feature: Check to see if a peer is up
23 maj 2012 kl. 14:35 skrev Alex Balashov: > That comes down to whether userspace, SIP stack-level OPTIONS pings are a > "good estimate of RTT". :-) Absolutely. But it's at least an estimate better than 500 ms in most situations. It does affect the quality of the call. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Quote for feature: Check to see if a peer is up
Just for the archives: Don't forget that you can use the SIPPEER() dialplan funciton to check the status of the peer with qualify=on before you place the call in the dialplan. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Quote for feature: Check to see if a peer is up
23 maj 2012 kl. 14:21 skrev Kevin P. Fleming: > While this behavior is technically not RFC3261 compliant (and I've had > discussions about it with at least one of the RFC's authors), it's quite > useful in making decisions about whether a peer has become unavailable more > quickly than would normally be possible. For a local peer that responds to > OPTIONS requests in 100ms or less, if that peer stops responding, Asterisk > will be able to make that determination in approximately 6 seconds, instead > of the 32 seconds that would normally be required. Form the RFC: "T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms." "The default value for T1 is 500 ms. T1 is an estimate of the RTT between the client and server transactions. Elements MAY (though it is NOT RECOMMENDED ) use smaller values of T1 within closed, private networks that do not permit general Internet connection. T1 MAY be chosen larger, and this is RECOMMENDED if it is known in advance (such as on high latency access links) that the RTT is larger. Whatever the value of T1, the exponential backoffs on retransmissions described in this section MUST be used." I can't see how this is not RFC 3261 compliant. We have a good estimate of the RTT and use it. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] The final Asterisk SIP Masterclass - Register now for beautiful Barcelona, Spain!
Friends, I've been running the Asterisk SIP Masterclass for many years now. It's time to run the last show - partly with new material. Compared with the very first Asterisk SIP Masterclass I would say that I've rewritten 90% of the material. That's what happens during the class. Students ask questions, you write a slide. The word changes, you write a slide. You realize you've been wrong, you delete or edit a slide. It's a moving target. The last one of these classes that I teach will be in Barcelona, Spain - June 11th to June 15th. * Why the last one? -- Things change and you need to follow. During the last couple of years I've been running many, many in-house trainings and workshops covering both Asterisk, SIP in general and Kamailio. There seems to be more demand for customized trainings that boost a team and help them move forward. I will continue with these trainings, as well as try to come up with other trainings that will run just a few times - more lab oriented possibly. * What is this class? --- From the sales material at http://www.avanzada7.com: "This class is focusing on building a scalable SIP realtime network. With a combination of theory and practical labs, you will learn how to setup and configure Asterisk and Kamailio - the Open Source SIP server - in a scalable enterprise or service provider network. We will go through various kinds of setups and give you insight in the design of real SIP networks with Asterisk running in enterprise and service provider networks. The teacher Olle Johansson, has many years of experience as an Asterisk developer as well as a community member of Kamailio.org. By spending a week with Olle, you will get a lot of insight into current and future features, bugs and implementation details in a way that's hard to get otherwise. Olle is a consultant working with architecture and implementation of large scale communication platforms based on the SIP protocol. He has experience from service providers, universities, call center platforms as well as enterprise solutions. With experience of Unix and TCP/IP networking for over 20 years, he has a lot of insight and knowledge, which he is using as a teacher." The class is a five day high level class. You will meet not only myself, but also other students that work with these tools and protocols, learn from them and work together to solve issues in the labs. You need to have a basic knowledge of Linux (how to start/stop applications, edit text files and build applications) and Asterisk. This class is starting at a high level with Asterisk. If you rather use FreeSwitch but want to learn Kamailio that is no problem. You will just have to endure a few slides on Asterisk - but many of the issues apply to FreeSwitch as well as other PBXs too. The cost is 3.200 Euro ex VAT. Companies outside of EU do not pay VAT as well as companies in EU with a VAT registration number. If you have any questions or want to register, feel free to contact me directly. Looking forward to seeing you in Barcelona! Best regards, /Olle --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] PROMO: US DID's
8 nov 2011 kl. 13:24 skrev Kevin P. Fleming: > On 11/07/2011 04:21 PM, Yaro Donchenko wrote: >> We have promotional offer for calling card companies. >> >> Virtual PRI (21 Channels) with 100 DID’s from anywhere in US could be >> random markets for only $250/month. > > PRI circuits only have 21 channels now? Inflation sucks. :-) Don't complain, Kevin. In some southern European countries they're down to 12. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Asterisk Training for developers - from idea to reality?
Hello friends, For a few years, I've been working as an advisor and teacher to several Asterisk platform developers. They work with AGI, AMI and the dialplan. Now I wonder if there are people interested in such a training class? I've done several in-house trainings like this. I would guess topics would be: - SIP & Asterisk introduction - Models: SIP Proxy, PBX, Feature server, IVR server - Asterisk concepts - Asterisk Manager - AMI - Asterisk Gateway Interface - AGI - DTMF control - IVRs - Call quality over IP - Call scenarios - Queues - Transfers - Normal bridges - Conferences We would go through practical concepts like call transfers, statistics, dialplan integration, testing and much more. The class won't cover a particular programming language, students are expected to be able to handle their development environment by themselves, I would help with the Asterisk integration. The result would propably be 70% training, 30% workshop based on student feedback, ideas and requirements. Any takers? It's not a formal registration, just feedback whether this sounds interesting for you or not. If you have additional ideas, please don't hesitate to add them. Answers off-list directly to my e-mail so we don't clutter the list. THanks. /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Patch needed: https://issues.asterisk.org/view.php?id=14239
6 feb 2011 kl. 08.31 skrev Dovid Bender: > Hi, > > $150.00 bounty to fix: https://issues.asterisk.org/view.php?id=14239 for > 1.8.X. Would like patch for 1.6.X as well. $200.00 if accepted by mantis. > What does "accepted by mantis" mean? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk freelance developers needed
23 jul 2010 kl. 10.24 skrev Olle E. Johansson: > Friends, > > I need to expand my network of Asterisk developers that can help Edvina with > small and larger projects, both related to SIP and to other parts of > Asterisk. If you are interested, please send me e-mail with your current > experience of Asterisk and your CV. If you have any questions, please don't > hesitate to call me. > > If you're based in Sweden, we are also looking for new employees to expand > our team that works with cool large-scale Open Source Unified Communication > projects. Please check http://edvina.net or contact me off list. > > As it is holiday season here in Sweden, please don't expect immediate answers > :-) > > Best regards, > /Olle Additional clarifications based on feedback: * I need people with Asterisk development skills, modifying or adding C source code to the base product for inclusion in the Open Source product. * I do not need help with Asterisk configurations or recommendations for alternative products :-) Thanks, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Asterisk freelance developers needed
Friends, I need to expand my network of Asterisk developers that can help Edvina with small and larger projects, both related to SIP and to other parts of Asterisk. If you are interested, please send me e-mail with your current experience of Asterisk and your CV. If you have any questions, please don't hesitate to call me. If you're based in Sweden, we are also looking for new employees to expand our team that works with cool large-scale Open Source Unified Communication projects. Please check http://edvina.net or contact me off list. As it is holiday season here in Sweden, please don't expect immediate answers :-) Best regards, /Olle --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] subject: 1.4 vs 1.6
24 feb 2010 kl. 15.45 skrev Chris Bagnall: >> Can someone please let me know if you have a such experience? >> Also, do you have any other negative or positive comments on 1.6 > > Most of our clients are still on 1.4 installs - I must admit largely because > of the time taken to rewrite the numerous AEL macros we've written over the > years into the new 1.6 layout (GoSub). > > We've not had any major issues with 1.4 that make an upgrade to 1.6 a > pressing necessity. Obviously you'll want to make sure you update your 1.4 > installs as and when new versions become available (and have been tested), in > the interests of security more than anything. Please observe that 1.4 is still a maintained release and the ONLY long-term-support version. The next version with long term support is the future 1.8. Security patches are still applied to all releases from 1.2. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] [asterisk-users] Important security alert: update your dialplans now!
While we continue discussing all possible solutions to this and build an expanding knowledgebase, I would like to repeat myself and kindly ask everyone that blogs, twitters, talks and teaches about Asterisk to please spread the word and the links. Later today, there will be an official Asterisk Security Advisory published on http://www.asterisk.org. Take this as an oppurtunity to repeat yourselves and make sure that no Asterisk admin anywhere talking any language can miss this information. If you are involved in a third-party project or commercial company that builds a GUI or anything that manages Asterisk dialplans, please make sure that your users are aware of this and that your project also releases information about this and if needed, updates your platform. If you are a distributor of Asterisk-related equipment, please don't hesitate to inform your resellers and customers. Let's use all the resources we have to make sure that we limit the damage that this issue causes. And thank you to all of you that has already been forwarding information about this issue. Best regards, /Olle My original alert is to be found here: http://lists.digium.com/pipermail/asterisk-users/2010-February/244721.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Phoen Museum videos
30 jan 2010 kl. 02.45 skrev Rehan Allah Wala: > Very interesting phone related videos, share by Moshe of net2phone via > Facebook. > > I wanted to share with u guys. > > It shows how far we have come, and where we can go from here > > http://www.techistan.net/phonemuseum/ > I just love those videos! Thanks Moshe and Rehan! And I think we need to shape up. Haven't seen telephony engineers properly dressed like the ones in that film at any Astricon. Only memory that I can recall is when Steve Sokol and myself showed up in a black suit at the very first Astricon and white shirt. Obviously we knew how to dress properly in this business! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Remote SIP monitor
6 jan 2010 kl. 15.30 skrev Jared Smith: > On Wed, 2010-01-06 at 09:45 +0100, Olle E. Johansson wrote: >> I'm adding manager events and storing data in a realtime database - >> one record per call leg. What I'm wondering is how we should handle >> call transfers and hold situations. A call that's transferred has >> multiple streams and RTCP is only valid for one stream. > > Would it make more sense to expose the RTCP information as part of the > CEL (Call Event Logging) infrastructure? That way, you could tell what > events in the call may have triggered the additional streams. In a > perfect world, we might even have something like: > > Event: Incoming call from Alice > Event: Outgoing call to Bob > Event: Asterisk bridges Alice to Bob > Event: RTCP report > Event: RTCP report > Event: RTCP report > Event: Alice places Bob on hold > Event: RTCP report (new stream, Bob hears hold music from Asterisk) > Event: Unhold > Event: RTPC report (Alice and Bob, again) > Event: Bob transfers Alice to Charlie > Event: RTCP report (new stream, Alice and Charlie) > Event: Hangup > > Obviously that's an oversimplified example, but I really think it makes > more sense to put the RTCP reports in the CEL logs, rather than having > another arbitrary log for the data. Maybe we should move this > discussion to the -dev list to discuss the more technical details? Well, since I'm paid to fix 1.4, CEL is out of the question at this point. When I move the code forward to trunk, CEL may be the proper way, depending on how people will work with CEL. The issue is that we get the final RTCP reports after or in close relationship to hangup.The CDR system can't handle that, and I don't know if CEL has improved this functionality. I'm right now considering the Shakespearian question "what is a stream?". RTCP has a stream identifier and when we end a stream, we should send an RTCP goodbye. Now, when we place a stream on hold, it's not ending a stream really. In Asterisk, we're stopping both rtp and rtcp when placed on hold, which is a bug. When we transfer, it's a new stream coming up. Interesting... > > Anyhoo, just wanted to add my own two cents (US cents, before interest, > taxes, depreciation, and amortization) Always appreciated! You know my paypal - o...@edvina.net :-) /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Remote SIP monitor
6 jan 2010 kl. 09.31 skrev Matt Riddell: > On 6/01/10 9:14 PM, Olle E. Johansson wrote: >> I would like feedback on what's missing in the AMI in order to monitor an >> Asterisk platform. I've added a lot of events and some actions to AMI in >> order to enhance monitoring, but new ideas are always welcome. > > RTCP :) > > That's probably it. Event: RTPQuality Privilege: call,all Channel: SIP/jarl.webway.se-0006 PVTcallid: 76a02abe54fbbacf23574acf0903a...@192.168.40.12 RTPmedia: audio RTPsendformat: ulaw RTPrecvformat: ulaw RTPlocalssrc: 1461473865 RTPremotessrc: 1365699479 RTPrtt: 0.00 RTPLocalJitter: 0.52 RTPRemoteJitter: 0.00 RTPLocalPacketLoss: 0 RTPRemotePacketLoss: 0 >From my pinefrog-1.4 branch :-) There's a lot to be done here. I'm adding manager events and storing data in a realtime database - one record per call leg. What I'm wondering is how we should handle call transfers and hold situations. A call that's transferred has multiple streams and RTCP is only valid for one stream. We should propably send one message like this per stream and have a counter. If you call someone locally and then transfers that person to me - the first call will be fine, but going from you all the way to the deep frozen north will means that a lot of packets just give up, freeze to death and die so that part of the call will be awful. We need to make sure we separate the issues and that we're somehow able to relate quality to defined peers in sip.conf. The RTCP engine in Asterisk needs an overhaul, and I got partial funding to do it. Sitting with wireshark, asterisk and five different phones and compare the RTCP traffic. Seems like the implementations vary on the phone side as well. > > I'd kinda like to be able to get memory, cpu, disk info, via the manager > because all up that's what I care about pretty much on most of the > nodes, but understand that they don't really have much to do with Asterisk. Well, we could have a res_system.so module that did that. > > Reverse Triggers would be nice :D > > Like, connect to port x on addr y if you get a certain event and send > some info. We do have that in voicemail. I was thinking about it for the named ACLs. Personally I think if we expose the information in manager and the security events log, a third party app could send snmp traps or nagios events. The res_snmp code is hard to get into and I don't think we have a trap API in there - but that's of course one standardized way of doing it. > That way you could have central daemons that log from end nodes. Yes. > > You can always connect back to the Manager and parse the info, but with > a lot of machines the load can get kinda high. > > If Asterisk were able to make an outbound connection you'd only see > traffic when something was up. Or an asterisk monitor running on that machine. I also made a dialplan app where you can check that a manager account is logged in. If it's not, if the monitor is missing, you can perform actions inside your asterisk now. It's not merged yet, waiting for approval in reviewboard, but I have it running in production. This is really an interesting discussion! /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Remote SIP monitor
6 jan 2010 kl. 02.56 skrev Matt Riddell: > On 6/01/10 1:28 PM, Erik Lagerway wrote: >> Do any of you know of a simple remote SIP monitor service or >> application/software? Looking for something I can point towards my >> various SIP servers to monitor registrations, ongoing call connectivity >> and email/SMS me if there is a problem etc. >> >> I don't want to install appliances or dedicate servers in the network >> for this, any ideas? > > Munin, Nagios, MRTG etc. > > I personally used to use MRTG, but am now using Munin. > > If you do decide to go down that path and need modules for Asterisk, let > me know I'm currently running: > > 1. Asterisk SIP Peers (online/offline/lagged) > 2. Asterisk IAX2 Peers (online/offline/lagged) > 3. Asterisk Concurrent Channels > 4. Num calls in last hour (answer/no answer/busy etc) - just from MySQL > CDR disposition. > Asterisk AMI - the manager interface - supports 1-3. You can also trigger alarms and get manager events when you reach maxcalls (setting in asterisk.conf). As you said, #4 is solved by a SQL query. I would like feedback on what's missing in the AMI in order to monitor an Asterisk platform. I've added a lot of events and some actions to AMI in order to enhance monitoring, but new ideas are always welcome. /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Need some used/refurbished or brand new IP Phones
30 dec 2009 kl. 15.03 skrev Kevin P. Fleming: > Peter Beckman wrote: >> On Wed, 30 Dec 2009, Alex Balashov wrote: >> >>> I'll start after you with this innovative new negative proposition >>> trend: I cannot modify the solar orbit of Earth, thanks. >> >> Crap! I was just gonna post and ask that... > > Finally, a place where I can safely post all the things I've learned I > cannot do. Just what I've been waiting for! > > 1. I cannot shoot fire from my fingertips. > 2. I cannot fail to remember that I am falling, and thus begin flying. > 3. I cannot stand the flavor of 'dailing minties'. Kevin, You better behave and wish all of these as New Year wishes. They might come true! /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem
29 nov 2009 kl. 17.47 skrev Mueller, Alexander: > 29.11. 17:37:19,060 Action: Originate > Channel: SIP/2000 > Context: originating > Exten: #*00123456798 > Priority: 1 > CallerID: 2000 > Variable: Outbound_CALLERID=07615987654321 > ActionID: ORIGINATE_452 Instead of SIP/2000 you can define local/ex...@context and in that extension add SIP headers and perform any magic you want. /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem
29 nov 2009 kl. 15.32 skrev Kevin P. Fleming: > Olle E. Johansson wrote: > >> Secondly, this mailing list is for Asterisk development - new code and >> issues with the Asterisk code. Your questions would get a faster and better >> answer on asterisk-users. Even if it's about development, it's about >> development of your own app, not about Asterisk. > > Actually, this is the -biz list :-) > Oh, one of those days. Then it's much much better to ask for advice on the asterisk-users list! Sorry. /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem
29 nov 2009 kl. 12.14 skrev Mueller, Alexander: > CS, that's my problem, exactly how you describe it: > but why can't you redirect the AMI Originate to a dialplan script that uses the SipAddHeader? > > After lots of experimenting, I now come from AMI into the dialplan by just > calling "Action: Command\r\nCommand: console dial #*2000" from the AMI, where > #* is the flag for originating (which is handled in the dialplan), and I'm in > the dialplan, so far so good. I can set the SIPHeader there an ring the local > phone (2000) via "Dial", the local phone takes the call automatically, fine. > > But then I have only one leg, I have not yet found out how to connect this > first leg inside the dialplan with the second leg. > > How can this originating be done, what has to come after Dial in order to > establish the outgoing call ? > > I have read the description of "Dial()" several times now, hard to > understand, it doesn't seem to offer a 2-leg-addressing (internal number and > external number). > > I also thought about creating call files in the "outgoing" directory, but > then again the problem with the missing SIPAddHeader() will lead to failure. > > Still unsolved ... > You have to learn about the local channel and how to use that from Originate. I think that's covered in the Asterisk book, downloadable from asteriskdocs.org and in many examples on the web. Secondly, this mailing list is for Asterisk development - new code and issues with the Asterisk code. Your questions would get a faster and better answer on asterisk-users. Even if it's about development, it's about development of your own app, not about Asterisk. Good luck in your Asterisk adventures! Regards /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Inhouse trainings for Asterisk, SIP and SIP Proxys
During the last year, I've had a number of inhouse training sessions covering * Asterisk * Asterisk and SIP integration * Asterisk application integration - AGI and AMI * SIP * SIP Proxys - OpenSER/Kamalio SIP-router I'm currently planning the training schedule for the spring of 2010 for our open classes that will run in London, Berlin, Stockholm, Orlando/FL and Burlington/MA. If you are interested in in-house trainings, please contact us now so that we can plan ahead. The standard classes that I run are * Asterisk Masterclass (similar to the bootcamp) * Asterisk SIP Masterclass - moving on from the bootcamp/dCAP level and focusing more on building scalable SIP solutions * SIP-router SIP Masterclass - For the OpenSER/Kamailio gurus-to-be The Asterisk and SIP-router Masterclasses are organized with Daniel-Constatin Mierla, Asipto - one of the core developers in SIP-router/OpenSER/Kamailio. In-house trainings are often more effective if you have a team that needs training for a specific project. I can freely pick presentations from my standard classes or make new ones so that the training is 100% adopted to your business needs. I have 15 years of experience doing trainings in various topics, from basic TCP/IP networking to PKI and IP-telephony. I wrote the first Asterisk Bootcamp and organized the trainings with Steven Sokol in the USA and in Europe, so I have long-term experience of doing training in Asterisk and VoIP too. We do organize trainings with local partners. If you are interested in co-organizing trainings with us, please don't hesitate to contact me. Looking forward to your reply! Best regards, /Olle --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP >From PBX to large scale implementations for carriers. Contact us today! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Asterisk 1.4 RTCP support - want to fund fixing it?
Friends in the Asterisk community. Recent testing has proved to me that the Asterisk 1.4 RTCP support, the protocol that measures media quality factors such as round trip time, jitter and packet loss, is not up to standard. There are many bug reports in the tracker about bad calculations, but the fact is that we're basing all RTCP stats on random data today, if anything at all. The RTCP packet sent from the remote end contains several information blocks. Our code assumes the first one is of one type and the next one is of another type, which is not correct. Thus it mixes sender and receiver reports. In trunk, RTCP maintains aggregated data about the packet loss and jitter for a call, something 1.4 does not do, it seems to report the last reported value for round trip and jitter. Asterisk should also parse the RTCP SDES information block and send one, as that block contains the overall session identifier for a call, the one that we should use to aggregate data at the end of the call to present the RTPAUDIOQOS etc variables. I am searching funding from Asterisk service providers, users and integrators to fix this. I want to fix 1.4 so it shows correct data per call and verify that trunk works correctly. 1.4 is in use today in many production environments and trunk is, hopefully, going to be the next long-term support Asterisk version (pending decision). Unless someone that provides funds wants me to check 1.6.x versions and provide fixes for them, I won't focus on those releases. Please contact me OFF LIST if you are interested in funding this work. If a few interested parties can fund one work-day each, so I can spend at least three full days on this issue, I think a lot can be done to improve our RTCP support. Thanks, /Olle --- * Olle E. Johansson - o...@edvina.net * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] OpenSER/Kamailio SIP Masterclass, Nov 9-13, Berlin
Hello, next edition of Kamailio SIP Masterclass takes place in Berlin, Germany, November 9-13, 2009. The training is focused on teaching Kamailio along 5 full days, up to advanced level, touching everything needed to build large and secure VoIP networks, integration with Asterisk media server as well as NGN-class of services for telephony and IP unified communication, such as instant messaging, presence, integration with social networking. Teachers: - Daniel-Constantin Mierla - founder and developer of Kamailio SIP server - Olle E. Johansson - Asterisk developer, consultant in large-scale SIP networks Registration details and brochure are available at: http://www.asipto.com/index.php/sip-router-masterclass/ A unique opportunity to learn everything about 2009's winner of InfoWorld's Best Open Source Networking Software. http://www.kamailio.org/mos/view/News/NewsItem/Best-of-Open-Source-Software-Awards-2009/ Please don't hesitate to write me e-mail for any question about this class. Early registration ending soon - secure your seat today to benefit of the early-bird-discounts! Best regards, /Olle --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] A hacker attack on asterisk
4 sep 2009 kl. 19.21 skrev Andy day: > Rehan, > > Asterisk is likely looking at the sip headers for IP authentication > and not > the actual IP headers. SIP headers can be spoofed, but I don't > believe they > can spoof the IP packets and still have it routed properly to this > customer > unless they are on the same network. If the customer does a packet > capture > (tcpdump tethereal etc) they should see the ip and sip headers do > not match > on those calls. They could use IP tables or some other ACL to block > the > hackers. There is a current bug in 1.6 for TCP connections (with or without TLS) that may be in action, where asterisk instead of looking at IP headers actually match on the Contact:. This is wrong and will be fixed soon in all 1.6 versions and trunk. For UDP, we actually DO look at the IP headers when we match incoming calls with peers. For user matching, we do match on the From: header. In addition we have authentication schemes for incoming calls for both users and peers. I do recommend ucing the ACL as well as authentication. /O --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Looking for a SIP Termination Provider
5 aug 2009 kl. 18.16 skrev Alex Balashov: > Olle E. Johansson wrote: > >> We can provide termination in remote parts of Törnskogen, Sollentuna, >> Sweden... :-) > > I heard the tin can and string access charges are stupendous! Would > you > like to enter into a profit-sharing agreement like no-good, risk-free > arbitrage players? :-) Yeah, the margins are enough for us to share and still afford the high- life of the Open Source and Jet Set communities. Welcome to my tin-can ´n´string business consortium! Btw, we're metric in the consortium. Only metric minutes charged. No more 60 secs a minute... /o ;-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Looking for a SIP Termination Provider
5 aug 2009 kl. 12.11 skrev Faiz Rehman: > We've been having problem in finding a really reliable company to > provide us with SIP Termination. So far we've used VoiceTrading > (Premium routes) > and have problems with calls not being connected, messages not being > played, false answer, etc etc. > Any help would be greatly appreciated. We can provide termination in remote parts of Törnskogen, Sollentuna, Sweden... :-) It would help if you explained where you need termination. You will get better answers to more specific questions. Please remember that this is a highly international mailing list. Cheers, /Olle PS. And if you want termination in Törnskogen, a string and a tin can is available outside of my office :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] PBX got Hacked
11 mar 2009 kl. 20.31 skrev Trixter aka Bret McDanel:
> On Wed, 2009-03-11 at 15:13 -0400, Andrew M. Lauppe wrote:
>>> Despite of all the arguments on other things we could do, why not
>>> increase
>>> the level of security in Asterisk if there is a possibility to do
>>> so?
>>>
>> Bottom line here, I think, is that the security holes aren't just in
>> Asterisk, they're in SIP, and Asterisk has to support SIP. It is SIP
>> that passes the usernames/passwords in plaintext. If SIP supported a
>> more secure authentication scheme, Asterisk would support it.
>>
>
> sip does do more secure auth, TLS but its not supported in asterisk
> because it requires TCP (RFC requires tcp support anyway, yet asterisk
> does not officially do that either).
>
> And passwords are NOT in plaintext.
>
> The username, nonce, and what you are doing (REGISTER for example) are
> all cleartext, but the password is not. The nonce is a short duration
> disposable number to prevent replay attacks.
To clarify: The password is never sent over the wire. It's a challenge-
response
authentication mechanism that sends a MD5 digest of a combination of
data sent in clear text and a shared secret that is not sent at all
("password").
I see two other alternatives:
- TLS authentication. This requires a lot of certificate/key pair
management.
- Stronger challenge-response by moving away from MD5 to SHAxxx.
/O
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Re: [asterisk-biz] PBX got Hacked
11 mar 2009 kl. 20.03 skrev Remco Barendse: > > Despite of all the arguments on other things we could do, why not > increase > the level of security in Asterisk if there is a possibility to do so? As always with Open Source, it's a matter of funding. We have worked out two rather detailed plans for new security architectures in Asterisk, but as with the chan_sip3 (pineapple) project, we've haven't got funding for the work. I do wish that more companies basing their business on Asterisk would step forward and fund development work in Asterisk in order to be able to create new functionality. /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] VON Europe Asterisk commun ity meeting
28 maj 2007 kl. 11.36 skrev S. A. Kamran: Hi, I am going to visit VON Europe and would like to meet Asterisk community there. Please email me off the list if you are visiting VON Stockholm or you are located in or around Frankfurt, Germany. There is an Asterisk meeting on the VON agenda. See you there! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Sponsorships for Codename Pineapple (chan_sip3) now open
What is Codename Pineapple? --- Codename Pineapple is a rewrite of the current Asterisk SIP channel. The goal is to make a SIP implementation that is more compatible with the current SIP specs and one that can participate better in a SIP network with proxys. Support for TCP/TLS, SIP transactions and other new SIP features are planned, as well as a change in the way we configure SIP devices. No more "friends", "users" and "peers". You can read more about Codename Pineapple on http://www.codename- pineapple.org A bit more technical information, including my todo-list, is in the doxygen docs: http://www.codename-pineapple.org/doc/html/chan_sip3_00index.html * Why sponsor? --- Development takes time, time costs money. As a free Asterisk developer, not being employed by anyone by myself, I need sponsors in order to be able to develop. For Codename Pineapple, I would like to focus for the most part of six months so that we can see progress. That won't happen without sponsorships, since in that case I would have to find other projects to work with and Pineapple would have to roll along on spare time. If I'm lucky enough to find sponsors for more than what I need, I will contract subcontractors to add to the speed of the project. * How do my company become a sponsor? - You sign up for sponsorship for a period of time. The cost depends on the level of sponsorship, but I'm of course open for suggestions on one-time fees too. Check the sponsorship web page on http:// www.codename-pineapple.org/sponsor.shtml Current sponsors are * Voop - the Internet Dialtone - http://www.voop.com * Nuvio - applications delivered http://www.nuvio.com * Digium - the Asterisk Company http://www.digium.com and of course Edvina. Don't hesitate to contact me if you have any questions or want to proceed with a sponsorship! Thanks for your support, /Olle * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk Logo
13 feb 2007 kl. 10.16 skrev AmberVoIP: Hello, Our company will be at Cebit and we want to use asterisk and openpbx logo, as most of our products are related to asterisk. I remember, same question was before, but not found at list. So, where i can get information about usage of Asterisk logotype and about legal issues? There's a pointer in the Asterisk LICENSE http://svn.digium.com/view/asterisk/trunk/LICENSE /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] SIP Chan Out Restrictions
31 dec 2006 kl. 13.12 skrev Rehan Allah Wala: i do not need an account, I need to restrict the call based on the called party id. Ie you own the did number 12126559343 and you buy from me 5 channels of incoming, I want to restrict the calls going from our end to your end without any sip user nor peer to channels = 5 Call forwarding services like virtualphoneline.com do not offer any sip users to restrict the calls qty. The user is identified based on the DNID number and the amount of channels the user has purchased for this DNID Why not use groupcount for this? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] SIP Chan Out Restrictions
31 dec 2006 kl. 07.14 skrev Rehan Allah Wala: > Rehan, > > Quoting Rehan Allah Wala <[EMAIL PROTECTED]>: > > > > > Hello, > > > > Has anyone implimented SIP and IAX out going channels qty > > restrictions on per user bases? > > You mean limiting usage on channels per user ? i.e. 5 channels for user > x and 10 channels for user b ? Yes, that's what I mean. > > > If so, I would be interesed in buying the code. > > Whats your bet ? :) Show it to me working and make me an offer, it has to be done via Chan_sip on asterisk code level. It's in chan_sip, called call_limit and available for free as open source. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Want to work in a cool group? Asterisk engineers needed in Norway and Sweden
On the behalf of a business partner I'm looking for more people to help us build a network based on Asterisk around Europe. Linux/Unix skills, Asterisk skills. Either good sysadmins that can help us install/configure/admin a LAMPA - (LAMP + Asterisk) or developers that can help us build support systems in C, PHP. We have both open permanent positions and short projects. Please contact me via e-mail if you're interested and I'll forward the information. You won't get a reply in return e-mail within seconds due to my workload (yes, we need more people) but you will get a reply. Best regards, /Olle --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Hospitality industry
23 aug 2006 kl. 21.22 skrev Erv Bauman: We are looking for Asterisk to RS232 for use with Property Management Systems in the Hospitality industry. Ideally, this would be a Mitel SX-200 emulation (or another PBX popular in this market) and would allow for data to pass to and from the PMS through the RS232 interface. This would allow for features required in the industry such as: Wake up calls Call accounting Room status Etc. That is interesting. Is the protocol over the RS232 interface standardized? Any documents to show how complicated this might be? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] from ip address variable ?
6 jul 2006 kl. 04.14 skrev Rehan AllahWala:
like oh323
http://www.voip-info.org/wiki/view/H323+Variables
Isnt there a variable for SIP
Like
${OH323_RADDR} - Contains the remote IP address/port of the
connection.
That's the very old way. We are moving towards dialplan functions
where possible.
Check the SIPPEER and SIPCHANINFO dialplan functions.
/O
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Re: [asterisk-biz] from ip address variable ?
30 jun 2006 kl. 03.22 skrev Rehan AllahWala: http://www.voip-info.org/wiki-Asterisk+variables Has any one written or know of a variable to capture the from ip address of the call from the headers in extensions.conf ? Depends on what protocol you are using. Check the SIP_PEER and IAX_PEER functions. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Coming trainings: Asterisk SIP Masterclass, Chicago, Asterisk Bootcamp at the Beach, Spain and Asterisk Bootcamp, Boston ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Register now for the Asterisk SIP masterclass and the European BeachCamp!
Just a short message from Edvina - the Asterisk Training company: The Asterisk SIP masterclass is in Chicago, Illinois next week. It's going to be a lot of labs and theory focused on how to make Asterisk work in a scalable SIP network. This class continues on an advanced level from the bootcamp class, so you need experience with Asterisk and Linux/Unix to attend. We still have a few seats and if you register directly to us, this first class is only USD 2.950. The next class will be priced USD 3.500. Today, we're launching the first ever BeachCamp - Asterisk Bootcamp in Southern Spain. A class where all students and teachers stay in the same hotel, close to the Mediterranean beach. The class will be held in English, but with a Spanish-speaking assistant teacher. It will be a lot of fun, a lot of Asterisk and a lot of teaching. At the end, we have the dCAP exams. The Asterisk BeachCamp is at the end of September, 25-29th, in Malaga, Spain and is organized in cooperation with Avanzada 7 - our Spanish training partner. For both classes, I will be the head teacher. We have had a lot of interest in the BeachBootCamp (based on rumours about it), so make sure you register early. * For more information, visit http://edvina.net * For registrations, send mail to [EMAIL PROTECTED] See you in trainings! Best regards, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] IAX2 based new products (subject change)
24 maj 2006 kl. 21.47 skrev Greg Boehnlein: On Wed, 24 May 2006, Paul wrote: So stir up some new products. I would love to see an iax2 ata that takes plugin fxo and fxs modules. It should also support iax2 trunking. That would allow more of us to support your efforts with our wallets. I second that request! :) An IAX2 ata has to have features not found anywhere else. So far, I haven't found anyone with support for RSA authentication and IAX2 encryption, two very important IAX2 features... Signed "Anonymous IAX2 user" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Asterisk Bootcamp in Europe :: June 12-16 and the Asterisk SIP Masterclass
** Asterisk Training :: Asterisk Bootcamp Our class in New York next week is already sold out and we're now planning the next batch of classes. We will soon release information about the ASTERISK SIP MASTERCLASS, to be held in Chicago July 10-14. More information further down in this mail. ** 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 well as more advanced configurations using E1/PRI, SIP and IAX2 protocols. Stockholm in June is a wonderful city with lots of activity, lots of sunshine and Asterisk techies :-) Facts about this training: Teacher: Olle E. Johansson, Asterisk developer and trainer. Material: Training slides (over 300 pages), The Asterisk Quick Reference Guide Dates: June 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 [EMAIL PROTECTED] 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 * Ed Guy, the architect behind Free World Dialup * Terry Wilson, a consultant that has built service provider SIP networks The class agenda is being worked on now, but will include: * Asterisk basics - a recap * SIP - an introduction to the protocol * SIP proxys and network infrastructure * The Asterisk SIP channel - introduction * Traversing firewalls and NAT devices * Key system functionality * SIP phones - audio and video * Building a SIP network with Asterisk and SIP proxys * SIP test tools As the bootcamp, this class will involve a lot of labs. At this point, we're opening up for early bird registration on this class. Since we have no product sheets, you are taking some chance, but will get a lower price. For registrations before June 15th, refering to this mail, you will get the class for 2950 USD (plus VAT in Europe). The regular price is 3500 USD. We will soon publish location information on our web site. See you on the trainings! Best regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Register for the Edvina.net Asterisk Bootcamp - May 22-26, New Jersey, USA
Just a quick note to say that we're quickly running out of seats for the Asterisk Bootcamp, May 22-26 in Edison, New Jersey, USA. For information about the bootcamp, please visit: http://edvina.net/training/bootcamp.shtm The trainers for this bootcamp will be Olle E. Johansson, Ed Guy and Terry Wilson. The price is USD 3.000. For European Union companies, VAT is added if we can't get your VAT registration ID. Please mail your registration and queries to [EMAIL PROTECTED] Our phone number is +46 8 96 40 20 Regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Need a small modification to Park Function
11 apr 2006 kl. 18.21 skrev BJ Weschke: On 4/11/06, Alexander Lopez <[EMAIL PROTECTED]> wrote: I need someone to code this as I do not have the time at this time. Scope: Add an argument to Park to be able to (s)pecify the parking space and (m)ute the annoucement of space number. IE Park(709|sm) would put the transferee into space 709, and not play the annoucment of what space the person is in. If a call is already parked fail the transfer. I understand that this is a pretty trivial code mod but I am beat as I just pulled an all nighter. It's not a huge undertaking, but it's also not that trivial. You need to make certain that the space you're asking to park something at is within the valid range of spaces and you need to make sure that you're not asking the system to park a call in a space that is already occupied. You could use the multiparking function for this and define a single space parking lot. It's in testing in test-this-branch if you want to test. One thing we don't have is the silent parking. That would have to be implemented as an option per parkinglot. Also, the SIP channel plays it for SIP transfers without using the Park() application, so that needs to be fixed somehow as well. I don't know if other VoIP channels have similar behaviour - or the Zap channel for that matter. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Meet Asterisk Europe http:// www.meetasterisk.com * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] MeetAsterisk in Europe - register today!
Friends, At the end of this month, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 * London May 5 * Stockholm May 19 (Close to Von Europe) MeetAsterisk is the one-day training that introduces Asterisk for a beginner, both from a business perspective and a technical perspective. You will get insights in how to use Asterisk in your business, as well as an introduction in how to install and set up Asterisk. It's a day filled with information to give you a quick-start with Asterisk. Find out the complete schedule at http://www.meetasterisk.com and register today! See you at MeetAsterisk! /Olle PS. MeetAsterisk will also contain a brief introduction to the new functions in the coming version of Asterisk - Asterisk 1.4 - to be released this summer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Astricon Training :: Now renamed to Edvina Asterisk Training! - Europe training next
Friends, The Astricon Training program that I have been teaching all of last year is now renamed and produced by my own company, Edvina AB in Sweden. We are working on a new schedule for the year and will soon have a lot of exciting news for you. At this point I wanted to alert you of the next bootcamp and dCAP exam: * Edvina Asterisk Bootcamp - Stockholm, Sweden - April 3-7, 2006 -- This is the original Asterisk Bootcamp, with myself as a teacher and the material that has been used and updated for many trainings. It's a tough one-week class that starts from the very basic and quickly moves to advanced configurations. You need to have some prior experience of installing software and managing a Linux/Unix system. All students have a PC and will install a fully working Asterisk PBX. During the week, we will build a business PBX configuration as well as more advanced configurations using E1/PRI, SIP and IAX2 protocols. You can bring your own hardware as well. Facts about this training: -- * Teacher: Olle E. Johansson, Asterisk developer and trainer. * Material: Printed Training slides (over 300 pages), The Asterisk Quick Reference Guide * Dates: April 3-7 (starting 10 AM Monday, ending noon friday * Options: dCAP exam friday afternoon, April 7th (ending around 5.30 PM) * Price: 2.500 Euro (ex VAT). 200 Euro (ex VAT) for dCAP. * All trainings are pre-paid. Register by e-mail to [EMAIL PROTECTED] today. All details are available on http://edvina.net/training/ After this class, we are planning classes in London and Boston this spring. We are also open for partnerships with resellers and distributors. Any questions? Please don't hesitate to e-mail us at [EMAIL PROTECTED] We can assist you finding a hotel, if needed. Looking forward to seeing you in Stockholm in the spring time! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Sponsor Asterisk development project: SIP attended transfers
trixter aka Bret McDanel wrote: On Wed, 2006-02-01 at 12:38 +0100, Olle E Johansson wrote: trixter aka Bret McDanel wrote: I think what you are saying, and please correct me if I am wrong, is that most of what you are doing in terms of these features is already done, and that its to work with the current chan_sip, basically just adding features to it. Is that correct? Yes, the work that this mail thread is about, is done on current chan_sip. Moving to another SIP library is something to be considered for a future version of chan_sip, in the chan_sip3 project. Now to ask what appears to me to be an obvious question. Again not to demean any work that you have done, nor the situation that it put you in, but if chan_sip3 would be a new sip stack and in theory obsolete your work by using a sip stack that presumably has all the features you described plus potentially additional ones (CNG, VAD, etc) why should resources not be put towards that project instead? I understand that this can sound inflamatory and that is not my intent, email often sounds different than it is intended, either too harsh or too jovial, my intent is just to gather information. Good question. Chan_sip3 is propably a year from now. I will look for sponsors for that development soon, but a lot of service providers and users need improved SIP transfer support sooner than that, in a stack that even with the flaws it has, is tested. Chan_sip3 will, knowing history, take a few development cycles to complete and be stable enough to run in a service provider or business production environment. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Sponsor Asterisk development project: SIP attended transfers
trixter aka Bret McDanel wrote: I think what you are saying, and please correct me if I am wrong, is that most of what you are doing in terms of these features is already done, and that its to work with the current chan_sip, basically just adding features to it. Is that correct? Yes, the work that this mail thread is about, is done on current chan_sip. Moving to another SIP library is something to be considered for a future version of chan_sip, in the chan_sip3 project. Apologies for my swenglish :-) /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Sponsor Asterisk development project: SIP attended transfers
trixter aka Bret McDanel wrote: On Wed, 2006-02-01 at 11:28 +0100, Olle E Johansson wrote: Even though it works in their environment, there is still some work to do to finish this quite large change and make it more generic and complete for standalone servers, as well as cleaning the source code up for peer review and possible commit. I need funding up to $10.000 USD to be able to dedicate time to complete this project and submit it to the Asterisk project. not to demean your work but why cant an existing sip stack be used like freeswitch.org is doing? There are good stacks out there that do all this, and as such it wouldnt take as much work becuase you wouldnt have to reinvent the wheel. That is a decision for the future, for chan_sip3. Most of this work is done and targeted at current chan_sip. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Sponsor Asterisk development project: SIP attended transfers
Last year, I spent a lot of time rewriting SIP transfers in chan_sip, in order to enhance the support for attended transfers, especially in the case where two servers where supported. This was paid for by a service provider, who after they installed it in their production systems decided not to pay all of the fees that we had agreed upon. Even though it works in their environment, there is still some work to do to finish this quite large change and make it more generic and complete for standalone servers, as well as cleaning the source code up for peer review and possible commit. I need funding up to $10.000 USD to be able to dedicate time to complete this project and submit it to the Asterisk project. The code * Makes transfers fail properly (right now, we send the transfer target a congestion tone or a busy signal, with no recovery) * Adds a scheme to allow/deny transfers per peer/user or in general * Adds support for INVITE with replaces headers * Adds support for attended transfers with call legs on different servers Most of this have been tested in depth, and is in production today. I already have one service provider contributing 1.000 USD towards this project. I will start as soon as I have guarantees of 10.000 USD, not to operate at a loss again with this project. Any contribution is welcome. If you are interested, please mail me off list. Thank you for your support! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Last call: Asterisk Bootcamp in Stockholm, Sweden
Just a short reminder: The next Asterisk Bootcamp and dCAP certification is in Stockholm, Sweden february 6-10. More information regarding prices, hotels and registration form is available http://www.astricon.net/training/ See you in Stockholm! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[Asterisk-biz] Astricon Training in Europe q1 2006
** Astricon Trainings in Europe - book now! We have scheduled two trainings for Europe in the first quarter of 2006: * The Asterisk Bootcamp - Feb 6-10 2006 The complete Asterisk one-week training including the dCAP examination. This training includes a lot of labs and in-depth knowledge about Asterisk * Introduction to Asterisk - March 6-8 A three day introduction to how to run Asterisk in your network. It's a good start for the person that is going to be managing Asterisk ast the office PBX. Both classes are in Stockholm, Sweden. Registrations are open now. http://www.astricon.net/training/ See you in Stockholm! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [Asterisk-biz] Looking for Asterisk Flavoured Fish
Matt Riddell wrote: > Will pay well for the delicate taste of Asterisk flavoured fish. > Now !! Anyone that knows help you will, or re-mail again. /O :-) ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
[Asterisk-biz] Astricon 2005 Network Sponsor wanted
During the two Astricon shows we had so far, network connectivity has been a hot topic. For Astricon 2004, the hotel promised us that they had a state-of-the-art network and that their provider's best network engineer would be on site to make sure that it worked. ...it worked for almost an hour... 500 computer users in one hotel was a good stress-test for both the network and the network engineer. For Astricon Europe 2005, the hotel had fibre and free Internet access. It worked for two days, which was a huge improvement. Now, for Astricon 2005, we're looking for outside help. We are allowed to set up our own network in the building. If there are companies out there that can provide us with a network - internet connection - that can handle all the traffic from the attendees, expected to be at least 750, we are willing to discuss an attractive sponsorship deal. We are going to be in the Hyatt hotel, a few blocks from Disneyland, in Anaheim, California. Contact us on [EMAIL PROTECTED] if you are willing to take this challenge! /Olle Johansson ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
[Asterisk-biz] Astricon Europe :: Register for dCAP exams now!
** Certify your knowledge of Asterisk in Madrid! There will be an oppurtunity to take the dCAP exam at Astricon Europe in Madrid, the Asterisk user's conference next week. We need registrations for dCAP in advance, so please sign up on the web site so we can plan the resources. Read more about the Digium Certified Asterisk Professional exam on http://www.astricon.net/europe/dcap.shtml See you in Madrid! ** Astricon Training in Denver, Colorado, July 2005 Last minute chance to register for the Astricon Training in Denver. Check our web site for more information and on line registration. http://www.astricon.net/training/ Regards, /Olle ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
[Asterisk-biz] Meet Asterisk in Stockholm!
Last chance to register for Meet Asterisk! In Stockholm on friday, may 27th. Meet Asterisk is a one-day introduction to Asterisk for new users and everyone interested in what Asterisk can do and can't do. Register today on http://www.astricon.net/meetasterisk ! Regards, /Olle ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
[Asterisk-biz] Astricon Europe :: OPEN for business
Astricon Europe - the Asterisk conference and exhibition June 15-17th this summer, is now open for registration. The web site is updated to accept registration as well as booking of hotel through an external agent that works with us. We will soon also release our spouse program, with events for the husband/wife/partner that follows you to Madrid and wants to explore Spain. * http://www.astricon.net/europe/ -- SPEAKERS WANTED During the coming week, I will start working with the conference program. I have many good proposals from speakers, but still have open slots both for tutorials and the conference programme. Send your proposal to me today! * http://www.astricon.net/europe/speakers/ -- Meet Asterisk! Stockholm May 27th Meet Asterisk! - the one-day introduction to Asterisk will be held in Stockholm next time. Registration is open and it's only 200 USD for a one-day comprehensive training. * http://www.astricon.net/meetasterisk/ -- Astricon Training: Seattle fully booked, Denver is open The five day Astricon bootcamp is quickly filling up. Seattle in April is now fully booked. We might have room, pending last minute cancellation. The Denver training in July is quickly getting booked. We are finalizing the schedule for the fall now. If you want to participate in Denver, book now. * http://www.astricon.net/training/ -- SPONSORING ASTRICON EUROPE 2005 If you want to sponsor Astricon Europe 2005 and participate in the exhibition, contact us now to receive the information material and price list. We have limited space in the exhibition. Send e-mail to [EMAIL PROTECTED] to reach us to get the process started. Get your logotype on the Astricon.net web site today! Best regards, /Olle and Steve ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [Asterisk-biz] copyright --may I use it for commercial business
nicknie wrote: Hi All, I'm very interested in Asterisk PBX and have successfully installed it in my home with 2 SIP phonesets. After further studying this system, I find its functions are very useful in small business office. My questions: 1. May I install it in my company office for internal use? Of course, it's open source. 2. If possible, may I do some further development to sell this system? Who should I pay for the license and how much? Please do further development and contribute it back to the community! As long as you are using it yourself, you don't have to pay a license. If you *sell* Asterisk with your own additions, you either have to give the source code of your additions away to the buyer under the GPL license and he can do whatever he wants to do with it - or buy a commercial license from Digium and thus get the commercial rights to the software and the right to sell your customized version. Regarding prices for that, contact Digium sales. http://www.digium.com Best regards, /Olle ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz
