SIP IP PBX | voip pbx | pbx system | sip | asterisk pbx | pbx phone | Free PBX - Directory VoIP Directory http://www.sipippbx.com/voip-provider-directory/ Sun, 05 Sep 2010 03:10:24 -0700 Sigsiu Online Business Index 2 FeedCreator Kylink http://www.sipippbx.com/voip-provider-directory/kylink.html NIPX-820 is the new mini IP-enabled PBX of Kylink Communication Corp. that integrates voice solution of PSTN and VoIP technology in one digital PABX. It provides various interfaces to connect with single line telephone, digital telephone, and also supports DISA, Caller ID, and VoIP features. NIPX-820 is the suitable communication solution especially for the SOHO use, with all-in-one box economical design, equipped with VoIP interface assisting you to step into the internet communication era linking the the world anywhere. Sun, 05 Sep 2010 03:10:24 -0700 OneBox http://www.sipippbx.com/voip-provider-directory/onebox.html Onebox Receptionist : The Onebox Receptionist Virtual PBX will answer your calls with a professionally recorded custom greeting and offer menu options for call routing. Calls can be sent to any employee, in any location, anywhere in the world. Unlike a standard PBX (phone system), there is no hardware or software to buy or maintain. All this at a very affordable, very flat monthly rate. Sun, 05 Sep 2010 03:10:24 -0700 FacetCorp http://www.sipippbx.com/voip-provider-directory/facetcorp.html FacetPhone includes all the normal features of a PBX plus much more as you can see below. With FacetPhone there are no feature packages to choose from. You always get all the features for one low price. Sun, 05 Sep 2010 03:10:24 -0700 Business VoIP http://www.sipippbx.com/voip-provider-directory/business-voip.html Business VoIP reviews and rates various VoIP plans from the perspective of a small to medium size business. There is also a constantly growing collection of articles on VoIP selection and implementation. (http://www.msn.com) Sun, 05 Sep 2010 03:10:24 -0700 Newport-networks http://www.sipippbx.com/voip-provider-directory/newport-networks.html The 1460 BC is deployed as part of a distributed SBC solution. The 1460 Border Controller can control one or more Border Gateways using a standard Megaco/H.248 interface. Physical separation of signalling and media allows centralised signalling control and distributed media control. The 1460 Border Controller (BC) enables peering and interconnect of VoIP signalling between operators. In addition it allows managed IP-based voice and multimedia signalling to be securely delivered to consumers and businesses. Key capabilities provided by the 1460 Border Controller are: The ability to traverse corporate, consumer and core network NAPT and Firewall devices for SIP signalling. Quality of Service enforcement via session admission control. Security protection for the core network, for customers, and for service revenue. Regulatory compliance providing Lawful Intercept and Emergency Call Handling. The 1460 BC can scale between 5,000 and 190,000 signalling sessions in a single, resilient 19 chassis; call processing performance can be scaled independently to achieve up to 1,000 calls per second. Sun, 05 Sep 2010 03:10:24 -0700 Quescom http://www.sipippbx.com/voip-provider-directory/quescom.html QuesCom 400 IP/GSM : GSM gateway for IP-PBX Immediate Tangible Savings Reduce the cost of landline to GSM calls without compromising quality thanks to VoIP GSM gateways. Technical specifications * First VoIP - GSM gateway * Digital Sound Quality * Integrated SMS gateway * Shared gateway architecture * Quick return on investment * From 2 GSM channell * Compatible with all H.323 or SIP IP-PBX * Compatible with GSM 900 or DCS 1800 * Powerful and flexible administration * Ethernet 10/100 Mbp * Codecs G.711, G.723.1, G.729a, GSM QuesCom 400 GSM : GSM gateway for ISDN PBX Not just a way to save money This GSM gateway for ISDN PBX does not only provide a huge telecom bill reduction but also a powerful set of tools thanks to optional packs : Sun, 05 Sep 2010 03:10:24 -0700 Polypix http://www.sipippbx.com/voip-provider-directory/polypix.html IP-2005 is IP Phone which is based on the standard SIP (IETF RFC3261). Major characteristics that differentiate IP-2500 from existing phones are 1) Voice Packet Encryption to prevent from eavesdropping, 2) network security on Kernel levels, 3) 3-Way Calling and 4) multiple calling (over 4 lines). Moreover, it is able to connect PCs using two Fast Ethernet Switch ports at the back like IP Sharers. In addition, IP-2500 enables text-chatting and video conferencing on PCs through interworking with SIP-based messengers. IP-2500 enables full-duplex (2-ways) media communication in private IP or NAT and Firewall environment without changing existing settings when it is interworked with SIP Media Controller Proxy. IP-2500 is designed to give a preference to voice data processing over PC data to secure excellent toll-quality. It is realized by hardware. In addition, high performance DSP algorithm is applied to reduce voice delay time on Jitter Buffering, Echo Cancellation, VAD and Audio Streaming. IP-2500 tested with other SIP products and services such as CISCO, VocalData, 3COM, Avaya and Net2Phone to maintain the interoperability of international standard SIP and have been continuously participated in official interoperability tests. Sun, 05 Sep 2010 03:10:24 -0700 Valid8 http://www.sipippbx.com/voip-provider-directory/valid8.html Valid8.com VoIP/NGN Test Benefits Affirms and enhances product performance * Rigorous, easy-to-use test solutions set standard for excellence * Dedicated senior engineering support for customers worldwide Decreases time-to-market * Highly flexible platform adapts easily for future testing needs * Rapidly-deployed, comprehensive testing suites support a broad range of protocols Increases confidence and competitive advantage * Complete testing solutions derived from extensive international experience in telecommunications validation and customized development * Validate new technologies to ensure highest level of performance testing solutions for Voice over IP (VoIP) H.323 * SIP * MEGACO * MGCP * SIGTRAN * RTP Sun, 05 Sep 2010 03:10:24 -0700 Artizanetworks. http://www.sipippbx.com/voip-provider-directory/artizanetworks..html Artiza VoIP Analyzer Artiza VoIP Analyzer is the software for analyzing VoIP-related protocols, (H.323, MEGACO(H.248), MGCP*, SIP*, etc.) *Scheduled which captures packets running on the local area network (LAN), by using PC and general-purpose NIC card. Artiza VoIP Multimedia Traffic Generator Artiza VoIP Multimedia Traffic Generator (MMTG) is powerful VoIP traffic generator. With a single MMTG, you can simulate up to 5000 SIP UA and 130 RTP Stream. Sun, 05 Sep 2010 03:10:24 -0700 VerticalCommunications http://www.sipippbx.com/voip-provider-directory/verticalcommunications.html Vertical Comdial MP5000® Business Communications System: Enabling Telecommunications to Keep Pace with Growing Enterprises - Economically The MP5000 is the ideal telecommunications solution for growing organizations with remote sites and road warriors, and with heavy intra-organizational calling and conferencing requirements. It’s easily administered from a single point and supports a broad spectrum of hard and soft endpoint mixes and advanced call handling applications. Add remote sites and users quickly and easily. Deploy advanced next-generation SIP and IP functionality, from peer-to-peer video calling to unified messaging, where it makes sense; preserve your existing telecommunications infrastructure where it doesn’t. Sun, 05 Sep 2010 03:10:24 -0700