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http://kb.smartvox.co.uk/


Welcome to the Smartvox Knowledgebase

test bench

Smartvox is a Limited company registered in the UK and specialising in the design and configuration of VoIP solutions primarily using open source products such as OpenSIPS, Asterisk  and FreeSWITCH. We have excellent test facilities and, as you can see in the photo, a nice selection of VoIP phones used for solution testing. The articles you will find on this web-based knowledge-base are original material derived from the results of internal research projects, most of which are self funded.

Smartvox is a small consultancy business and you are welcome to call and ask for John to discuss your IP Telephony requirements. We can provide assistance with any of the following:

  • Consultancy for OpenSIPS and Asterisk projects (configuration – setup – support)
  • Design and setup of resilient VoIP solutions for Internet Telephony Service Providers
  • Installation, configuration and support of Asterisk-based business telephone systems
  • Selection and supply of VoIP and networking hardware, including the Cisco small business range

Why not get in touch with us to discuss your requirements now – Smartvox Contact Details

The technical articles on this site are aimed at VoIP professionals and other users of open source VoIP products. They are organised in categories for ease of navigation. Use the Category headings shown on the right, the tabs along the top of the page or the search facility to find the information of interest to you. I update old material and try to add several new articles each year so come back often to see what’s new.

At the end of each article you will find options to post comments or questions. Even if you are in a hurry, using the coloured voting buttons and the Facebook Like button is very quick. Your feedback is encouraged and is most welcome.

Happy browsing.

John Quick, Smartvox Limited, February 2013.

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2013.04.06
22:21:22 (*.160.42.88)
엮인글 :
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List of Articles
번호 제목 글쓴이 조회 수sort 추천 수 날짜
82 Presence Tutorial OpenXCAP setup admin 42119   2014-08-18
 
81 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 41889   2014-08-13
 
80 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 41634   2014-08-24
 
79 A lightweight RPC library based on XML and HTTP admin 41598   2014-08-18
 
78 Real-time Charging System for Telecom & ISP environments admin 41512   2014-08-23
 
77 A Survey of Open Source Products for Building a SIP Communication Platform admin 41505   2014-10-18
 
76 Opensips install debian admin 41501   2014-03-03
 
75 opensips-1.10.0_src.tar.gz experimental source code documentation admin 41091   2014-03-09
 
74 SigIMS IMS Platform admin 41028   2014-05-24
 
73 [Sipdroid] SIP data collection study tour admin 40854   2014-08-23
 
72 UAC Registrant Module admin 40775   2014-09-28
 
71 Multimedia Service Platform admin 40485   2014-03-06
 
70 MediaProxy Installation Guide admin 40435   2014-08-10
 
69 CANCEL MESSAGE not handled correctly admin 40413   2014-08-23
 
68 OpenSER_from_an_asterisk_POV file admin 40361   2013-04-06
 
67 OpenSIPS Kick Start‎: VIDEO admin 40105   2013-02-20
 
66 A2Billing and OpenSIPS config admin 39887   2014-10-20
 
65 OPENSIPS EBOOK admin 39720   2014-08-21
 
64 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 39719   2014-08-12
 
63 fusionPBX install debian wheezy admin 39358   2014-08-09
 
62 opensips.cfg for Asterisk admin 39237   2014-10-20
 
61 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 38858   2014-08-11
 
60 opensips NAT Traversal Module admin 38815   2014-10-02
 
59 opensips Nat script with RTPPROXY - English Good perfect admin 38589   2014-08-15
 
58 kamailio.cfg configuration Example admin 38567   2014-10-04
 
57 OPENSIP Training VIDEO admin 38419   2013-02-20
 
56 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 38033   2014-08-23
 
55 OpenSIPS as Homer Capture server admin 37987   2014-08-13
 
54 Kamailio Nat Traversal using RTPProxy admin 37847   2014-08-11
 
53 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 37841   2014-08-12