- freeswitch-biz@lists.freeswitch.org - FreeSWITCH Mailing Lists

[Freeswitch-biz] Fake Calls Block using FS
by Yogesh Patel 26 May '13

26 May '13

Hi We are looking for solution to prevent Fake calls being terminated to our VOIP Gateways. Our requirement is to build up a dynamic blacklist/white list which can automatically detect fake generated calls either by the number format pre assigned. (i.e. 93701111xxx) or by number of call attempts from a specific ANI OR to specific DINS (i.e. 93702222222 hitting to GW more then 5 times in 1 min.) all those calls should be rejected from the FS before they hit to voip gateway. Currently calls are passing thru our switch directly to the gateway and we are manually blocking such numbers, however we are looking for solution to automatically detect the number and block them if they are not pre configured to block. Let me know if you have any query. Please advise if any one has such solution readily available or can develop and if so how much it will cost. Thank you, Yogesh Patel CEO YoipCarriers Yoip Networld Solutions Inc. * 1.647.896.7489 7 1.647.827.9313 MSN: yogesh(a)yoipcarriers.com Skype: yoipcarriers www.yoipcarriers.com <http://www.yoipcarriers.com/>

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[Freeswitch-biz] Kamailio training in San Francisco, June 24-27
by Daniel-Constantin Mierla 10 May '13

10 May '13

Hello, I would like to announce the next edition of Kamailio Advanced Training in North America, taking place in San Francisco during June 24-27. Among topics are dynamic routing, scalability and high availability, solutions that allow building LCR engines, load balancers or scale existing VoIP platforms, gateways or media servers farms. Kamailio and FreeSwitch are known to interconnect very well, several online tutorials are already available out there authored by me. More details about the training can be found at: - http://asipto.com/u/katu In addition, if someone is interested in consultancy services at your premises in the area or just meeting for a drink, drop me an email. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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[Freeswitch-biz] Multiple Companies
by Jerome LEGRAND 02 May '13

02 May '13

Bonjour, Je débute dans le monde de freeswitch, mais venant d'elastix, je connais le monde la VOIP. Après beaucoup de temps passé sur l'installation de freeswitch sur ubuntu avec fusionPBX, je cherche à configurer mon système. Dans l'interface de fusionPBX, je peux rajouter facilement des extensions et cela fonctionne en appel interne. Cependant ce que je recherche, c'est l'utilisation de context multiples ( multi-tenant ? ). C'est a dire pourvoir avoir plusieurs entreprises avec des extension 1xxx chacune sans pouvoir contacter l'autre. Malgré beaucoup d'essais et tuto, je n'ai pas réussis ... Avez-vous des pistes ou recherches sur le sujet ? Merci d'avance ! Cordialement, Hello, I started in the world of FreeSwitch, but coming from elastix, I know the world of VOIP. After much time spent on installing FreeSwitch on ubuntu with fusionPBX, I try to configure my system. In the interface fusionPBX, I can easily add extensions and it works internally call. But what I want is the use of multiple context (multi-tenant?). That is to say several companies have filled with 1xxx each extension without access to the other. Despite many tests and tutorial, I did not succeed ... Do you have any leads or research on the subject? Thank you in advance! Sincerely, Jérome LEGRAND [cid:image001.png@01CE4748.481EAB70]Adoptez l'éco-attitude.N'imprimez ce mail que si c'est vraiment nécessaire ! AVIS DE CONFIDENTIALITE : Ce message et toutes les pièces jointes sont établis à l'intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d'en avertir immédiatement l'expéditeur. Toute utilisation de ce message non conforme à sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. Le réseau internet ne permettant pas d'assurer l'intégrité de ce message, ULTIMEO décline toute responsabilité au titre de ce message, dans l'hypothèse où il aurait été modifié.

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[Freeswitch-biz] Seeking help with recording overlapping
by Eli Finkelman 08 Apr '13

08 Apr '13

Hi, We're looking for someone to help us diagnose issues where 1 out of 100 or so recordings are overlapping. These recordings are being reviewed by a quality and performance measurement team, so they must be properly aligned. Please contact me directly at eli(a)teltechcorp.com with your rates and experience in dealing with this type of issue. Thank you in advance. Regards, Eli

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[Freeswitch-biz] Skypopen- insmod ./skypopen.ko
by Anouar Abrik 26 Mar '13

26 Mar '13

Hi I know this has been discussed a number of times and I did try going through every fix you had in every ticket but to no avail. I started off with a Ubuntu 12.04 LTS, knowing that a desktop distro won't work I installed a server distro, 12.04 LTS. I then proceeded to add a Kubuntu desktop and followed every step on that wiki page successfully till "How to build skypopen.ko OSS sound driver (required)", not that i did not install freeswitch or skype on kubuntu but on ubuntus terminal(ctrl+alt F1). None of the SND_OSS mods were loaded, I am fairly sure about that, yet when ever I try to run the insmod command it would end up with a resource busy error. I have a corei5 with 4 GB RAM and dual booting my linux and windows 7. I apologize but I am fairly new to this and I have a project for my bachelors that rests on your mod, any help/advice would be highly appreciated. -- Regards, Anouar Abrik *Graphics Designer* Direct Number(PK): +92-336-229-4557 Skype: donnymaniac * * * *

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[Freeswitch-biz] Seeking Technical Help for Quality Issues
by Avi Marcus 16 Feb '13

16 Feb '13

I'm interested in hiring someone to help identify and track down quality issues, for just a few hours a month when issues come up that I can't handle. I've been running a calling card and residential VoIP line for over a year. There are sporadic quality issues - DTMF, claims of crackling, dropped packets, out of sync audio, etc. Also, for the residential voip I rely on the SPA-2102 and it's generally been pretty good. However, when things don't quite work (e.g. no call waiting not working, constant reboots, losing connection) the device doesn't seem that helpful in debugging. I'm wondering if there's someone that can take a look at the CDRs, PCAPs, logs, etc and help figure out what's going wrong, especially analyzing RTP in PCAPs. This seems to be a skill much different than the basic config and routing issues I'm familiar with... Please contact me directly via email if this could be a fit. Thanks, -Avi Marcus BestFone

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[Freeswitch-biz] need hired help finding audio loss problem
by Royce Mitchell III 11 Jan '13

11 Jan '13

I've been fighting with this problem for over a week. I'm losing audio when calling devices on the AT&T wireless network via an Adtran Total Access 908e 2nd Gen. I have proof that audio is being sent to the FreeSWITCH, but FreeSWITCH's recordings of the calls drop audio whenever I stop hearing it on the phone. I need to hire an expert who can look at what my system is doing, find a configuration or help develop a patch ( maybe identify a new NDLB flag that's needed ) to get my setup working. Please contact me at the email address below, thanks. -- Royce Mitchell IT Consultant ITAS Solutions royce3(a)itas-solutions.com

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[Freeswitch-biz] Seeking Freeswitch developer - ASAP
by Anthony Caiozzo 04 Jan '13

04 Jan '13

FS Developers - Telchemy manufactures software that measures speech quality in real-time and in-situ from the end-user perspective. We are looking to retain a crackshot Freeswitch developer to help us enhance FS in a way that would be compatible/useful with our server side analytics platform. Initial project scope has been completed - we anticipate the complete project to take ~200 hours. Timing on this project is now and has budget allocated. Please contact me if you are interested or would like more information. -anthony Anthony Caiozzo Telchemy - www.telchemy.com <blocked::http://www.telchemy.com/> m: 617-312-5189 f: 678-387-3008 e: anthony.caiozzo(a)telchemy.com <blocked::mailto:anthony.caiozzo@telchemy.com> support: 1-866-TELCHEMY Skype: acaiozzo

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[Freeswitch-biz] The Edvina SIP Masterclass v2.0 in Miami FL - Dec 10-14 2012
by Olle E. Johansson 30 Oct '12

30 Oct '12

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 info(a)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 - oej(a)edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 * http://edvina.net/training/

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[Freeswitch-biz] FreeSWITCH / C Developer
by Travis May 29 Jun '12

29 Jun '12

FreeSWITCH / C Developer Company Background Founded in 2002, TollFreeForwarding.com (TFF: www.TollFreeForwarding.com<http://www.TollFreeForwarding.com>) is a technology driven, international telecommunications provider based in Los Angeles, California, with headquarters located near LAX airport. The company is privately owned and closely held. TFF has always been profitable and debt-free. Since the founding of TFF ten years ago, the company has grown expansively. With 40 employees, a new suite of services is under development that will change the way businesses handle customer communications through a communications centric, SaaS-based CRM. There are now several senior and mid-level openings in marketing, sales, software engineering, product development and programming for candidates that are looking to work in an innovative, technology-focused, fast paced, customer service-centric environment. Job Scope The ideal candidate is someone with deep experience in, and design with C programming, FreeSWITCH and TCP/IP Architecture. S/he participates as a high level technical expert in design, development, coding, testing, and debugging of new software. S/he works within a team to understand problems with software and develops specifications to resolve them. The ideal candidate is competent to work at the highest technical level of all phases of applications programming activities. The candidate will be responsible for programming in C, understanding C++ and FreeSWITCH modules. This position requires an individual who is able to produce well-written, well-documented, high-quality, and fast code. Excellent verbal and written English communication skills are a must. The final candidates will have the unique opportunity to build a world-class, communications-centric, SaaS-based CRM using the best technology and coding techniques available. This position is essential to supporting this company's rapid, sustainable and profitable growth. We are looking to fill this assignment immediately. Required Skills, Qualifications and Motivations * Natural programming and architecture abilities combined with deep expertise * 2+ years customizing FreeSWITCH and/or module development * 4+ years hands-on C development experience * Experience with developing cross platform applications using Apache Portable Runtime (APR) and OpenSSL * Excellent interpersonal, verbal, and written communication skills * Thrive on best practice coding without delaying projects due to over-optimization * Ability to self-test code and know weaknesses of chosen method * Open to code review and constructive criticism * Significant experience utilizing and creating 3rd party APIs * Experience with Linux Desired Skills, Qualifications and Motivations * Experience with C++ * Experience with libamf * Experience with mod_rtmp * Experience with Flex/ActionScript Education, Experience and Benefits * The successful applicant will have a Bachelor's Degree in Engineering, Computer Science or a related field. * Competitive salary * 401K with employer matching and immediate vesting * Health, Dental, Vision, Disability, & Life Insurance * Education Benefits To Apply * Send your PDF/Word resume to HR(a)TollFreeForwarding.com Travis May, Founder and CEO, TollFreeForwarding.com [Description: Description: cid:image001.gif@01CABBC1.33E972E0]<http://www.tollfreeforwarding.com/>

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