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nexVortex
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USA
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Monday, 2:32 PM EST by
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Thread started: Monday, 2:32 PM EST
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nexVortex is a telecommunications service provider focused on the business customer. We deliver the cost and technical advantages of Internet based network services without compromise in quality or service reliability. Take advantage of SIP Trunks to connect your IP PBX to the world and save money. nexVortex SIP Trunking provides: * Outbound calling to anywhere * Inbound calling with US numbers * Tollfree numbers * Emergency Services
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SIP Specific Firewalls - or better - Controllers
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Firewalls
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Feb 3 2010, 3:55 AM EST by
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Thread started: Feb 3 2010, 3:55 AM EST
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Leave your 'pure' Data protection in place - in fact don't mess with it at all. Go for a SIP Security Controller that: - understands and protects against the specific attacks that VoIP and SIP connections are subject to; - handles real time comms better than a traditional firewall; - is a separate unit that copes with everything the tradition firewall doesn't want to be bothered with; - performs SIP level checks and manipulation so that 'compatible' systems and connections actually connect!
We've always managed to iron out the idiosyncrcies of SIP trunks with the inflexibility of PBXs - so take a look at SIP Security Controllers at www.UM-Labs.com
cheers Andrew
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Free Asterisk Technical Support for all SIP Trunking Services
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SIP Trunk providers
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Jan 20 2010, 5:32 PM EST by
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Thread started: Jan 20 2010, 5:32 PM EST
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Need help? IP Communications provides free Asterisk Technical Support with all SIP Trunking Services. Simply signup for any of our SIP Trunk Services and we will assist you in configuring it with your Asterisk PBX.
For a limited time, Free Asterisk Support is available for our Free DIDs (www.ipcomms.net/product-freedid.html) .
IP Communications http://www.ipcomms.net http://www.ipcomms.net/product-freedid.html
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Ingate Repacement
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Discussion Forum
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Jan 6 2010, 1:29 PM EST by
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Thread started: Dec 9 2009, 6:23 PM EST
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Hi,
We are currently using an Ingate Siparator. Does anyone know of a product that will do the same as the Inagte but not as expensive?
Ideally the proxy would be an ethernet version only rather than DSL.
Thanks
Gavin
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Last Reply:
RE: Ingate Repacement
By: ,
Jan 6 2010, 1:29 PM EST
Hello, I've literally just signed up - so hope my post is in context. I am working with UM LABS and can recommend their SIP SECURITY CONTROLLER. - www.um-labs.com The RC-2100 handles up to 10 concurrent calls and starts at under GBP1,000. Happy to discuss, cheers Andrew
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NGTLive voip solution provider
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NGTLive Pc-Phone dialer
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Dec 14 2009, 2:01 AM EST by
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Thread started: Dec 12 2009, 7:59 AM EST
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Hi,
NGTLive Softphone/dialer: ==================================
Our highly integrated and full featured Softphone software is an advanced and next generation communication application that offers the clients a number of basic and advance phone utilities including mobile wireless telephony, VPN remote access and automatic phone dialing. We have designed our Softphone in compliance with client’s business processes and requirements to demonstrate an ultimate voice quality which supports G729, G723, CLI support and Echo Cancellation facilities. STANDARD SOFTPHONE FEATURES: 8 lines Call Hold/ Unhold Call Pickup Call Transfer(Xfer) Call Forward Do Not Disturb or DND Missed Call Indicator Call Redial/ Auto Redial Call Conference PC to Phone & phone to PC calling HTTP Authentication SIP UDP utility Codec Negotiation Large Address Book Call Time Display Credit Balance Display Mute Voice Mail Notification Speaker (with Ringer Device Selection) Auto Accept Call Full Duplex Audio Recording Real-Time Account Balance Display Balance calculation and display while the calling Time Reminder Display Call History PSTN connectivity using SIP gateways & soft-switches Audio tuning wizard
TECHNICAL FUNCTIONALITIES: ================================= Acoustic Echo Cancellation Packet concealing Packet Lost Concealment (PLC) Comfort Noise Generator(CNG) Silence Suppression UDP, TCP and TLS for SIP transport Voice Activity Detection Automatic Voice Gain Control (AGC) for voice Realm Settings Resampling Registration Timeout STUN Server Support ICE Support SIP Business Log Multiple Accounts SIP Message Log
mailme: ashish.dubey@ngtlive.com MSN: ashishdubey1981@hotmail.com
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Last Reply:
RE: NGTLive voip solution provider
By: ,
Dec 14 2009, 2:01 AM EST
This is not an item for discussion - it's and Advert and being so, it's in the wrong place. Please create a page in the Directory and NOT post adverts in the discussion area as they will be delete in the future.
Thanks you, Moderator
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MERA Systems To Present Solutions For Mobile Carriers at Asian Carrier
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The SIP School ~ News
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Dec 10 2009, 3:18 AM EST by
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Thread started: Sep 8 2009, 5:30 AM EDT
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MERA Systems, CBoss and Telejet will discuss current trends in mobile carrier market during workshop at Asian Carriers’ Conference.
Toronto, ON, September 8, 2009 – MERA Systems announces today it will hold a workshop to explore mobile services current trends and new opportunities for mobile market players emerging from development of core networks towards packet technologies at the 5th Asian Carriers’ Conference in Cebu, Philippines held on September, 9-12, 2009. CBoss and Telejet, leading VoIP billing developers, will join a workshop to contribute with their expert views and innovative solutions. “Today the market of mobile services is the fastest growing segment of telephony services. It is extremely attractive for carriers that operate in other segments. At the same time fixed line services are still widely used, and mobile carriers aspire to penetrate into this market,” – says Konstantin Nikashov, CEO of MERA Systems, - “MERA Systems has developed a set of solutions that comprise FMC, MVNO and IMS features to cover current needs of mobile market players and allow for cost effective, rapid and hassle free implementation.” MERA Systems experts will also highlight the demand for cost-effective and comprehensive VoIP solutions and discuss recently announced MERA Integrated solution and its target markets.
MERA Systems invites all concerned to join the workshop on September, 11 at 1:00 pm - 5:00 pm at Rosal Ballroom 3.
For more information visit www.mera-systems.com
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Why is It This Way
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Free Training
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Dec 10 2009, 3:16 AM EST by
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Thread started: Jan 23 2009, 7:41 PM EST
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What I want to know is why is it so hard for people don't chage their phone provider PSTL over to the internet Phone Service Provider?
Dose any one know the reasion ?
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Last Reply:
RE: Why is It This Way
By: ,
Dec 10 2009, 3:16 AM EST
Most people don't understand the concept and don"t forget this
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amir1983 |
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SIP server and bandwidth
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Discussion Forum
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Dec 10 2009, 3:12 AM EST by
amir90p |
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Thread started: Dec 3 2009, 10:51 AM EST
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Hi guys,
Sorry if my question seems a little weird. I'm totally new to VOIP and SIP! I want to know if a SIP server acts as a central server to transfer VOIP packages from/to users, or it's just a registration server. I mean for example suppose that user A is in Asia, user B is in Europe and the SIP server is in the US. Now, when A calls B, does it mean that ALL voice traffic goes from Asia, to US and then to Europe, or just from Asia to Europe? The problem I have is that I want to make my VOIP environment somehow that when A calls B, it uses direct connection to B. I don't want the SIP server to play any role in this voice connection other than registering that A & B are talking. Is it possible? This is because my SIP server have very limited bandwidth :(
Thank you in advance for your help
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Cisco 7960 to Mitel 3300
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The SIP School™ WIKI and Directory
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Dec 8 2009, 7:56 AM EST by
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Thread started: Apr 24 2009, 2:55 PM EDT
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I'm trying to set up a Cisco 7960 as a SIP client to a 3300. Was able to locate the How-to from Mitel but having trouble locating the SIP software. Anyone done this?
Dry Aquaman
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Last Reply:
RE: Cisco 7960 to Mitel 3300
By: ,
Dec 8 2009, 7:56 AM EST
Mitel came out with a good How-To. I was able to get everything going. Not worth the effort for what you get, however.
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NGTLive class 5 switching platform
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SIP Software
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0 |
Dec 1 2009, 7:51 AM EST by
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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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ashishdubey1981 |
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NGTLive encryption pc-phone and mobile dialer solution
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SIP Phones
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Dec 1 2009, 7:49 AM EST by
ashishdubey1981 |
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Thread started: Dec 1 2009, 7:49 AM EST
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Encryption Server: ==================
Encryption server is designed to handle very high traffic more than 10k concurrent calls through
Encryption enable client: ==========================
1. PC-Phone Dialer 2. Mobile Dialer A) Windows Mobile B) Symbian 3. Devices A) Encryption enable plug and play devices B) Desktop tunnel application for existing devices
For further details and for demo of the solution please contact following,
Thanks and Regards Ashish Dubey Business Manager(International Sales) 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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NGTLive class 5 switching platform
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SIP Hardware
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0 |
Dec 1 2009, 7:48 AM EST by
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Thread started: Dec 1 2009, 7:48 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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NGTLive Mobile dialer with encryption integrated
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SIP Trunk providers
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Dec 1 2009, 7:46 AM EST by
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Thread started: Dec 1 2009, 7:46 AM EST
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Hi,
NGTLive has launch mobile solution with encryption for all blocked countries.
Now NGTLive introduces combo offer in mobile solution with encryption ,
Go for NGTLive symbian mobile dialer with encryption and get windows mobile dialer customized as combo pack!!!!!!!
For more information and free demo please contact at,
Thanks and Regards Ashish Dubey Business Manager(International Sales) 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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Predictive Dialer,Call center SOftware,IVR
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Telecom Services
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Nov 25 2009, 2:45 AM EST by
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Thread started: Nov 25 2009, 2:45 AM EST
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Drishti provide Predictive Dialer,Call center SOftware,IVR etc.
If you have any questions about any of the above technologies or are interested in setting up a contact center, please feel free to get in touch.
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AQuA - Audio Quality Monitoring for Linux IP PBXs
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Linux
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Nov 24 2009, 8:31 AM EST by
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Thread started: Nov 24 2009, 8:31 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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AQuA - Audio Quality Analyzer for IP PBXs
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SIP Software
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0 |
Nov 24 2009, 8:28 AM EST by
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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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Broadvox SIP Trunking
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USA
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Nov 18 2009, 2:10 PM EST by
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Thread started: Nov 18 2009, 2:10 PM EST
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Founded in 2001, Broadvox is a worldwide leader in providing integrated managed VoIP services to SMB, enterprise, and carrier customers. It has deployed one of the largest full-featured global VoIP networks and is trusted by more than 200 telecommunications carriers, ASPs, ISPs, and over 3,000 businesses to transport over 10 billion minutes annually. The Broadvox Network Operations Center operating 24x7 provides the reliability, security, and quality of service required by the world’s most discriminating customers. Broadvox offers SIP Trunking, SIP origination and termination services, and hosted communications solutions. Broadvox is headquartered in Dallas, Texas. For more information about Broadvox, visit www.Broadvox.com.
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Free SIP Trunking From IPComms- Now even Better!
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IP Communications
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Nov 18 2009, 1:09 PM EST by
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Thread started: Nov 18 2009, 1:09 PM EST
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We are happy to announce that our Free SIP DIDs have gotten even better! We now make connecting to IPComms even easier by adding SIP Registration for all of our Free DIDs! This means you can test using a Softphone such as x-lite or zoiper. This also allows customers with Asterisk servers on dynamic ip address not have to worry about updating their ip address information as their settings change. To make it even better we are now allowing 10 mins of free outbound calling so that you can get everything on your system tested. We hope these new changes will help the community to continue the development of Asterisk and help with new users getting their services up and going with less pain. If you have any questions or can suggest any new features we should add please contact us.
Best Regards
Donald Hansil IP Communications PH: +1.678.460.1475 Fax: +1.678.868.1606 Email: Donald.Hansil@ipcomms.net NOC Email: noc@ipcomms.net Support: www.myipcomms.net
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Broadvox Wins Award for GO! SIP Trunking
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SIP Trunk providers
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Nov 17 2009, 1:34 PM EST by
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Thread started: Nov 17 2009, 1:34 PM EST
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Founded in 2001, Broadvox is a worldwide leader in providing integrated managed VoIP services to SMB, enterprise, and carrier customers. It has deployed one of the largest full-featured global VoIP networks and is trusted by more than 200 telecommunications carriers, ASPs, ISPs, and over 3,000 businesses to transport over 10 billion minutes annually. The Broadvox Network Operations Center operating 24x7 provides the reliability, security, and quality of service required by the world’s most discriminating customers. Broadvox offers SIP Trunking, SIP origination and termination services, and hosted communications solutions. Broadvox is headquartered in Dallas, Texas. For more information about Broadvox, visit www.Broadvox.com.
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Seeking for Egypt mobile, Germany mobile, Haiti mobile routes
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Discussion Forum
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Oct 14 2009, 10:44 AM EDT by
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Thread started: Oct 14 2009, 10:44 AM EDT
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Seeking for Egypt mobile, Germany mobile, Haiti mobile routes
--------------------- e-mail: alexander@speedflow.com ICQ: 101-476-107 MSN: alexander@speedflow.com
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