{"id":46,"date":"2006-04-13T15:40:58","date_gmt":"2006-04-13T19:40:58","guid":{"rendered":"http:\/\/blog.voipsupply.com\/wordpress\/?p=48"},"modified":"2016-01-29T17:05:03","modified_gmt":"2016-01-29T17:05:03","slug":"digium-announces-new-hardware-products-at-von","status":"publish","type":"post","link":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/digium-announces-new-hardware-products-at-von\/","title":{"rendered":"Digium Announces New Hardware Products at VON"},"content":{"rendered":"<p><strong>Digium Announces New Hardware Products at VON<\/strong><br \/>\nNew Transcoder and Echo Cancellations Cards Improve Call Quality and Communication between PSTN and <a title=\"voip gateways\" href=\"\/voip-gateways\" onclick=\"ga('send', 'event', 'voip-insider-blog-post', 'click', 'VoIP Gateways');\">VoIP Gateways<\/a><\/p>\n<p>HUNTSVILLE, AL and SAN JOSE, CA \u2014 (March 14, 2006) &#8211; <em>Source: Digium Press Release<\/em> \u2014 Digium Inc., the creator of Asterisk\u2122, and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for <a title=\"free voip\" href=\"http:\/\/www.smithonvoip.com\">VoIP<\/a> and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1\/E1\/J1\/PRI cards with onboard hardware echo cancellation.<\/p>\n<p>&#8220;Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP,\u201d said Mark Spencer, president of Digium. &#8220;Not only are we constantly striving to improve Asterisk\u2019s performance, but we also want to contribute to the overall VoIP experience, while keeping costs low.\u201d<\/p>\n<p>The TC400P provides hardware transcoding of VoIP codecs; decreasing Asterisk&#8217;s work load and providing improved CPU efficiency and an increase in channel density over a software-only solution. The TC400P provides Asterisk with full transcoding support and hardware acceleration for the G.723.1 and G.729A codecs.<\/p>\n<p>The TE420P and TE415P improve upon Digium&#8217;s existing <a href=\"\/ip-pbx-hardware\/pci-cards\" onclick=\"ga('send', 'event', 'voip-insider-blog-post', 'click', 'TE411P');\">TE411P<\/a> and <a href=\"\/networking\/network-accessories\" onclick=\"ga('send', 'event', 'voip-insider-blog-post', 'click', 'TE406P');\">TE406P.<\/a> These premium interface cards provide carrier-grade echo cancellation, Voice Quality Enhancement (VQE), DTMF decoding, and tone recognition. Both cards minimize loads on the processor and PCI bus, and are designed to perform in the most difficult environments. The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels.<\/p>\n<p>Digium designed these solutions to be fully compatible with existing software applications and is completely integrated with the Asterisk platform. Additionally, the open source driver supports an API interface for custom application development. All new solutions support industry standard telephony and data protocols, including Primary Rate ISDN (both North American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC and Frame Relay data modes.<br \/>\n<strong><br \/>\nAbout Digium<\/strong><br \/>\nDigium is the original creator and primary developer of Asterisk, the industry&#8217;s first <a title=\"open source pbx\" href=\"\/ip-pbx-hardware\" onclick=\"ga('send', 'event', 'voip-insider-blog-post', 'click', 'open source PBX');\">open source PBX<\/a> and Asterisk Business Edition, the professional-grade version of Asterisk. Used in combination with Digium&#8217;s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures.<\/p>\n<p>Digium provides quality hardware and software products that enable telephony applications including legacy <a title=\"pbx topics\" href=\"http:\/\/www.pbxtopics.com\">PBX<\/a>, IVR, auto attendant, next generation <a title=\"gateways\" href=\"\/voip-gateways\" onclick=\"ga('send', 'event', 'voip-insider-blog-post', 'click', 'gateways');\">gateways<\/a>, media servers and application servers. Digium also offers a full range of professional services including consulting, technical support and customer software development services.<br \/>\n<strong><br \/>\nAbout Asterisk<\/strong><br \/>\nCode for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed to from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports US and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.<\/p>\n<p>The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.<\/p>\n","protected":false},"excerpt":{"rendered":"<p>Digium Announces New Hardware Products at VON New Transcoder and Echo Cancellations Cards Improve Call Quality and Communication between PSTN and VoIP Gateways HUNTSVILLE, AL and SAN JOSE, CA \u2014 (March 14, 2006) &#8211; Source: Digium Press Release \u2014 Digium Inc., the creator of Asterisk\u2122, and pioneer of open source telephony, today announced the availability [&hellip;]<\/p>\n","protected":false},"author":7,"featured_media":0,"comment_status":"open","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[1075,1218,1225],"tags":[],"class_list":["post-46","post","type-post","status-publish","format-standard","hentry","category-asterisk","category-open-source-voip","category-voip-hardware"],"_links":{"self":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/46","targetHints":{"allow":["GET"]}}],"collection":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/users\/7"}],"replies":[{"embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/comments?post=46"}],"version-history":[{"count":2,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/46\/revisions"}],"predecessor-version":[{"id":149453,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/46\/revisions\/149453"}],"wp:attachment":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/media?parent=46"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/categories?post=46"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/tags?post=46"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}