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| Started By | Thread Subject | Replies | Last Post | ||||
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| oksenka | Streamco Smartswitch - 4/5 class switch with built-in billing | 0 | Jun 8 2010, 8:38 AM EDT by oksenka | ||||
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Thread started: Jun 8 2010, 8:38 AM EDT
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Hello!
We offer you Streamco Smartswitch – high-performance and user-friendly software system for telephony calls processing. It’s a hybrid Class 4\5 solution that can be used as Softswitch, PBX, Voice Access Server, etc. Key Characteristics: - Built-in billing in real-time mode - Graphical interface with drag-and-drop for call handling configuration - Managment and reporting via web-interface - VoIP protocols support (SIP and H323), transcoding between them - wide codecs support (including G711, G723, G729, GSM, ILBC and others), transcoding between them - T38 fax support (pass-through) - Graphical interface with drag-and-drop for call handling configuration The system fits well in following business applications: - Wholesale Switching - Calling Cards - Call Termination - Residential Subscriber - Direct Inward Dialing (DID) - PBX - Branches Interconnection - SMS-based services - Telemarketing Do you want to learn more? Look here http://www.streamco.org.ua/ Best regards, Oksana Bolbot Streamco Sales manager sales@streamco.org.ua http://streamco.org.ua/
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SIP Software
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| Voipex | Ensuring your VoIP deployment is Robust, with good QOS and efficient | 0 | Apr 8 2010, 12:07 PM EDT by Voipex | ||||
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Thread started: Apr 8 2010, 12:07 PM EDT
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ViBE is a product desigend to ensure that deploying VoIP is as resilient and dependable as tradditional Voice was
ViBE delivers; 500% more calls through the same bandwidth with NO LOSS of Quality, Supports both Voice and other Data Applications over the same bandwdith, Failover on Hardware, Failover to alternative bandwidth with no loss of call, Bonding of multiple types of bandwdiths, Encryption, The Best QOS available, Real Time Stats for the underlying network not just calls in progress, RAIN mode which allows duplication of calls to ensure elimination of any intermittent jitter and instaneous failover with the client not even aware it has happened, |
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| ashishdubey1981 | NGTLive class 5 switching platform | 0 | Dec 1 2009, 7:51 AM EST by ashishdubey1981 | ||||
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Thread started: Dec 1 2009, 7:51 AM EST
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Hi,
VPeer is aimed specifically at Service Providers looking for upto 3,000 Concurrent Calls Class 4/5 VoIP Solution with a strong Reseller Panel, allowing them to host a pre-integrated Platform to accelerate their penetration into high-margin VoIP market. VPeer provides the shortest time-to-market to the Service Providers. Services that can be rendered through this Solution are: 1. Wholesale VoIP 2. Retail Net Telephony 3. VoIP Calling Card 4. International Call Back 5. Class 5 Voice over Broadband 6. Hosted PBX / IP Centrex For more information,please contact at Thanks and Regards Ashish Dubey www.ngtlive.com Phone: +17753130316 mailme: ashish.dubey@ngtlive.com MSN: ashishdubey1981@hotmail.com ashish.dubey@ngtlive.com Yahoo: ashishdubey52001@yahoo.co.in Skype: ashish.dubey1981
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Keyword tags:
callback
calling card
class 5 switch
retail voip
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| sevanaoy | AQuA - Audio Quality Analyzer for IP PBXs | 0 | Nov 24 2009, 8:28 AM EST by sevanaoy | ||||
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Thread started: Nov 24 2009, 8:28 AM EST
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AQuA - Audio Quality Analyzer
AQuA provides precise and extensive audio quality testing that can be easily brought to your company's own hardware enabling your monitoring server to graphing MOS/PESQ values for various voice termination locations. Many companies are using this product to test and monitor voice and audio quality of conference bridges, carriers and VoIP links. Typically companies that provide telecom solutions have automated "test dialers" to test networks of conference bridges, carriers, terminations etc, and the system is able to place test calls and record them to for example a wav file. And the only thing that is missing to obtain full test sequence is an ability to automatically compare wav files to determine if there are any significant audio quality problems. AQuA voice quality solution is designed exacty to fit these needs. In case of Asterisk for example voice quality monitoring using AQuA is implemented in the following way: 1. Originate a call on the monitoring server using Asterisk manager interface to a server which is running an echo application. 2. Monitor both inbound and outbound legs of the call and save them as wav files. 3. Use AQuA (Linux or Windows version of the software) to compare the wav files. This approach is quite effective as one does not need to purchase additional hardware thus utilizing existing infrastructure. We sincerely believe that many other companies will find AQuA product useful, competitive and cost effective solution for audio quality testing and monitoring. Thank you very much for your time and we are looking forward to hearing from you! Best regards, Endre Domiczi Sevana Oy CEO/Co-founder Phone: +358 9 23164165 Mobile: +372 53485178 http://www.sevana.fi
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| hankkarl | ISDN, R2, RBS to SIP interworking module for IP PBXes | 4 | Oct 5 2009, 10:14 AM EDT by hankkarl | ||||
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Thread started: Apr 20 2009, 10:21 AM EDT
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PSTN-SIP module allows IP PBXes to connect to the PSTN via ISDN PRI or BRI, QSIG, CAS R2 or RBS, or analog telephony. Pre-ported version available for Linux (Windows is on the roadmap) that interfaces to DAHDI (was Zaptel) boards so you can use the least expensive hardware solution. DAHDI over ethernet is also supported.
This product allows you to implement an ISDN (or other PSTN) interface with minimal knowledge of ISDN or SIP. Asterisk and other FOS PBXes benifit from gaining the ability to use the industry-proven TeleSoft ISDN stack that is not performance limited in the way that the free stacks are. Configuration and installation is much easier than configuring and installing the free stacks.
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| hankkarl | High Performance, High Quality, Fully Supported SIP Source Code | 0 | Apr 20 2009, 10:13 AM EDT by hankkarl | ||||
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Thread started: Apr 20 2009, 10:13 AM EDT
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High Performance, High Quality SIP source code written by experts in the Telecom field is fully supported. This stack was written with ease of use and portability in mind. It has both low CPU cycle and low per-call RAM requirements, and scales well so it is suitable for use in the largest SoftSwitches, conference bridges, B2BUAs and other devices as well as being suitable for small devices like TAs and Handsets. Various user interfaces are available, including the native API which is very powerful, and a softphone client for use on mobile devices which allows engineers to implement IMS SIP without a great knowledge of SIP.
Also available is an interface to the industry-standard TeleSoft ISDN and R2/RBS CAS stacks for use in gateways and interworking devices. http://www.nine-9s.com/prod_embedded_sip_protocol_stack.htm |
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| Anonymous | Softswitch&Billing/CallingCards/Sale/Rent | 1 | Feb 6 2009, 2:29 AM EST by Anonymous | ||||
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Thread started: Feb 5 2009, 6:36 AM EST
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IXC can provide you with a high-class INTEGRATED SoftSwitch and Billing – within one platform!
IXC Softswitch & Billing Features: - Authorization and routing of incoming traffic depending on IP and prefix simultaneously (that is not allowed by Cisco); - Switching h323 and SIP calls; - Codec Converter; - Handle interaction with Gatekeepers; - Handy system of problematic destinations closing; - Online active calls viewing; - Traffic monitoring using statistical characteristics like call duration, success call ratio, traffic values; - Quality of connection based on analysis of RTP session between terminal devices; - Different detailed elaboration of statistics; - Accounting of cross traffic flow between companies; - Handy generation of tariffs and possibility of calls cost converting if tariff changes had place; - Four-stage system of traffic flow routing including Least Cost routing; - Generation of reports for CDR checks; - View of routing tables both for a chosen customer, and for all customers together; - Work with deposit customers, possibility of calls breaking when balance is empty; - Work with time-depending tariffs Start your Calling Cards Business with our software! IXC company can provide you with Turnkey Solution for Calling Cards VoIP business calls origination and sending them into the world which includes IXC PrePaid Platform. The whole bunch for Calling Cards business is not so big as it may seem. Moreover, we provide 24/7 technical support!! You can purchase our software at once or take it for rent from us! We are looking forward to cooperation with you. Alex Sharafanenko IXC sales manager, Email: s.alex@ixc.ua msn: s.alex@ixc.ua office: +38044 2399740 icq: 320504689
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| dgolyanina | MERA Systems VoIP softswitches | 0 | Jan 30 2009, 6:45 AM EST by dgolyanina | ||||
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Thread started: Jan 30 2009, 6:45 AM EST
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Reach higher with the best VoIP softswitch ever!
MERA Systems is a respected vendor of VoIP software solutions for telecommunication operators. Its award-winning portfolio contains the following Class4/5 products. For wholesale operators: o MERA VoIP Transit Softswitch (MVTS) - VoIP softswitch with the gatekeeper and proxy functionality o MERA VoIP Transit Softswitch II (MVTS II) - All-in-one VoIP software solution with billing capabilities o MERA VoIP Transit Softswitch Pro (MVTS Pro) - Distributed VoIP softswitch with unmatched productivity For retail operators: o MERA IP Centrex - Flexible hosted VoIP solution for service providers Find out more at http://www.mera-systems.com
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IP Centrex
Mera
softswitch
VoIP
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| shotone | Pangean's SIP-based Global Intercom for PC and Pocket PC | 1 | Dec 9 2008, 1:58 AM EST by vocale | ||||
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Thread started: Dec 8 2008, 4:26 PM EST
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TMS's Product of the Year for 2007 and 2008, Pangean offers a "true" SIP Global Intercom for hoot 'n holler, conference and "walkie-talkie". The application requires a Windows Server and a multicast enabled network for group communications. The system is currently in use at NASA, Homeland Secuity and numerous brokerage firms for instant communications. Features include IM, Presence and Private Communications. System uses multicast for group communications and unicast for private group conference.
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hoot intercom broadcast
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| Anonymous | IXC SOftswitch FREE DEMO! | 0 | Dec 3 2008, 4:50 AM EST by Anonymous | ||||
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Thread started: Dec 3 2008, 4:50 AM EST
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IXC suggests you a completely authentic Free BSD based solutions for any type of VoIP business.
Our Softswitch has a user-friendly web-interface for you to manage working with it without any problems. Reliable software you can trust. IXC can offer you the following solutions for such VoIP services: 1. Wholesale / Retail Termination 2. Calling Cards Solution - PrePaid / PostPaid 3. Callback Solutions - ANI/SMS/WEB 4. IP-Phone Solution 5. PC2Phone Solution 6. Corporate Clients Solution 7. Last Mile Solution or if can be provided by a complete complex turnkey solution which includes more than one exact IXC VoIP solution. Also we offer 24/7 technical support ! You can buy it at once or take for rent at first. Contact us and try our DEMO version for FREE! For more detailed info browse our web-site at: www.IXC.ua Alex Sharafanenko IXC sales manager, Email: s.alex@ixc.ua msn: s.alex@ixc.ua office: +38044 2399740 icq: 320504689
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Softswitch
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| Anonymous | Award winning SIP Softswitch from Dialexia.com | 0 | Nov 27 2008, 4:51 PM EST by Anonymous | ||||
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Thread started: Nov 27 2008, 4:51 PM EST
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Dial-Gate is a carrier-class VoIP softswitch with billing solution for voice and data applications. The softswitch application is based on open standards such as SIP and provides a rich, highly customizable communication platform. Its programmable routing intelligence and support of network-calling plans can be assigned to endpoints in order to filter calls. Dial-Gate translates standard telephone numbers into their corresponding IP addresses and looks for available VoIP gateways or domains to place calls through to the PSTN. It functions as a softswitch by giving an integrated control point for various telephony services. The Dial-Gate billing solution provides real time monitoring, pre/post paid billing, rate/route table management, and CDR report generation. Dial-Gate is fully integrated with MS Exchange supplying the user with voicemail unification, instant messaging and video communication.
Contact Sales @ #300-181 Hymus Blvd Pointe-Claire, Qc, Canada H9R 5P4 Tel: +1-514-693-8500 www.dialexia.com |
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| Alex-IXC | VOIP Complete Solutions | 0 | Oct 17 2008, 7:12 AM EDT by Alex-IXC | ||||
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Thread started: Oct 17 2008, 7:12 AM EDT
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Dear Businessmen!
If you want to start any VoIP business, IXC (Interexchange Carrier) will willingly help you with its VoIP Software Solutions. IXC can offer you the following solutions for such VoIP services: 1. Wholesale / Retail Termination 2. Calling Cards Solution - PrePaid / PostPaid 3. Callback Solutions - ANI/SMS/WEB 4. IP-Phone Solution 5. PC2Phone Solution 6. Corporate Clients Solution 7. Last Mile Solution or if can be provided by a complete complex turnkey solution which includes more than one exact IXC VoIP solution. Also we offer 24/7 technical support ! You can buy it at once or take for rent at first. Contact us and try our DEMO version for FREE! For more detailed info browse our web-site at: www.IXC.ua Alex Sharafanenko IXC sales manager, Email: s.alex@ixc.ua msn: s.alex@ixc.ua office: +38044 2399740 icq: 320504689
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VoIP
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| Anonymous | SIP IVR | 0 | Sep 28 2008, 4:25 PM EDT by Anonymous | ||||
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Thread started: Sep 28 2008, 4:25 PM EDT
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Check out a low cost vxml 2.1 based sip ivr server from IEC - http://www.iec-sw.com
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sip ivr vxml voip
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