|
|
|
Cisco 7960 to Mitel 3300
|
The SIP School™ WIKI and Directory
|
3 |
Today, 5:54 PM EST by
|
|
|
Thread started: Apr 24 2009, 2:55 PM EDT
Watch
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
out of
found this valuable.
Do you find this valuable?
Show Last Reply
|
|
Last Reply:
RE: Cisco 7960 to Mitel 3300
By: ,
Today, 5:54 PM EST
Have you tried using X-Lite o connect to your Mitel? That will connect with minimal settings, at least you will know that the 3300 is OK.
out of
found this valuable.
Do you find this valuable?
|
|
|
amir1983 |
|
SIP server and bandwidth
|
Discussion Forum
|
0 |
Thursday, 10:51 AM EST by
amir1983 |
|
|
Thread started: Thursday, 10:51 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
NGTLive class 5 switching platform
|
SIP Software
|
0 |
Tuesday, 7:51 AM EST by
|
|
|
Thread started: Tuesday, 7:51 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
ashishdubey1981 |
|
NGTLive encryption pc-phone and mobile dialer solution
|
SIP Phones
|
0 |
Tuesday, 7:49 AM EST by
ashishdubey1981 |
|
|
Thread started: Tuesday, 7:49 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
NGTLive class 5 switching platform
|
SIP Hardware
|
0 |
Tuesday, 7:48 AM EST by
|
|
|
Thread started: Tuesday, 7:48 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
NGTLive Mobile dialer with encryption integrated
|
SIP Trunk providers
|
0 |
Tuesday, 7:46 AM EST by
|
|
|
Thread started: Tuesday, 7:46 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
Predictive Dialer,Call center SOftware,IVR
|
Telecom Services
|
0 |
Nov 25 2009, 2:45 AM EST by
|
|
|
Thread started: Nov 25 2009, 2:45 AM EST
Watch
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.
out of
found this valuable.
Do you find this valuable?
|
|
|
|
AQuA - Audio Quality Monitoring for Linux IP PBXs
|
Linux
|
0 |
Nov 24 2009, 8:31 AM EST by
|
|
|
Thread started: Nov 24 2009, 8:31 AM EST
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
AQuA - Audio Quality Analyzer for IP PBXs
|
SIP Software
|
0 |
Nov 24 2009, 8:28 AM EST by
|
|
|
Thread started: Nov 24 2009, 8:28 AM EST
Watch
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
1
out of
1 found this valuable.
Do you find this valuable?
Do you?
|
|
|
|
Broadvox SIP Trunking
|
USA
|
0 |
Nov 18 2009, 2:10 PM EST by
|
|
|
Thread started: Nov 18 2009, 2:10 PM EST
Watch
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.
out of
found this valuable.
Do you find this valuable?
|
|
|
|
Free SIP Trunking From IPComms- Now even Better!
|
IP Communications
|
0 |
Nov 18 2009, 1:09 PM EST by
|
|
|
Thread started: Nov 18 2009, 1:09 PM EST
Watch
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
1
out of
1 found this valuable.
Do you find this valuable?
Do you?
|
|
|
|
Broadvox Wins Award for GO! SIP Trunking
|
SIP Trunk providers
|
0 |
Nov 17 2009, 1:34 PM EST by
|
|
|
Thread started: Nov 17 2009, 1:34 PM EST
Watch
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.
out of
found this valuable.
Do you find this valuable?
|
|
|
|
Seeking for Egypt mobile, Germany mobile, Haiti mobile routes
|
Discussion Forum
|
0 |
Oct 14 2009, 10:44 AM EDT by
|
|
|
Thread started: Oct 14 2009, 10:44 AM EDT
Watch
Seeking for Egypt mobile, Germany mobile, Haiti mobile routes
--------------------- e-mail: alexander@speedflow.com ICQ: 101-476-107 MSN: alexander@speedflow.com
out of
found this valuable.
Do you find this valuable?
|
|
|
|
Can't Find it
|
My SIP Switch
|
0 |
Oct 14 2009, 4:10 AM EDT by
|
|
|
Thread started: Oct 14 2009, 4:10 AM EDT
Watch
Do you actually offer an installation of MySIPSwitch for public use like it says here? I can't find it, can you you give a link to it?
out of
found this valuable.
Do you find this valuable?
|
|
|
|
ISDN, R2, RBS to SIP interworking module for IP PBXes
|
SIP Software
|
4 |
Oct 5 2009, 10:14 AM EDT by
|
|
|
Thread started: Apr 20 2009, 10:21 AM EDT
Watch
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.
out of
found this valuable.
Do you find this valuable?
Show Last Reply
|
|
Last Reply:
RE: ISDN, R2, RBS to SIP interworking module for IP PBXes
By: ,
Oct 5 2009, 10:14 AM EDT
Unfortunately, Telecom has a history of companies claiming to embrace open standards and then adding their own "value added features" to differentiate themselves from other companies. Naturally, having these "differentiators" mean that the companies no longer provide a standard product.
Its understandable that the telecom carriers want to avoid having a commodity product.
I suspect that the VoIP carriers will first try to provide a standard product, then when they find they are constantly being underbid by companies willing to lose money, they will add differentiators, just as traditional telecom carriers have.
out of
found this valuable.
Do you find this valuable?
|
|
|
|
|
Why is It This Way
|
Free Training
|
1 |
Oct 1 2009, 10:47 AM EDT by
|
|
|
Thread started: Jan 23 2009, 7:41 PM EST
Watch
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 ?
1
out of
2 found this valuable.
Do you find this valuable?
Do you?
Show Last Reply
|
|
Last Reply:
RE: Why is It This Way
By: ,
Oct 1 2009, 10:47 AM EDT
Most people don't understand the concept, and when people don't understand something, they ignore it. Also there are a lot of people that can't get the bandwidth where they live to support it properly. It is up to us to get people to understand it, and get them using it. Thanks Dave C.
out of
found this valuable.
Do you find this valuable?
|
|
|
|
|
grnVoIP provides quality wholesale voip termination
|
SIP Wholesale Termination
|
1 |
Sep 27 2009, 6:22 PM EDT by
|
|
|
Thread started: May 29 2009, 9:38 AM EDT
Watch
GRNVoIP maintains relationships with over 50 quality A to Z VoIP termination carriers, including all Tier 1 VoIP carriers throughout the world.
View our rates: http://rates.grnvoip.com Visit our homepage: http://www.grnvoip.com
* Completely Automated - make calls immediately upon sign-up * Pay using credit card or Paypal * Low minimum purchase amount of $50 * SIP, H.323, G729, G711 supported * No Set-up Fees or monthly charges * Live In Minutes * 24x7 Tech Support * Control your account with online account management area. * Two route choices - select best quality or best rates
out of
found this valuable.
Do you find this valuable?
|
|
|
|
MERA Systems Will Officially Launch MERA Retail & Transit Unit at VON
|
The SIP School ~ News
|
0 |
Sep 17 2009, 7:44 AM EDT by
|
|
|
Thread started: Sep 17 2009, 7:44 AM EDT
Watch
MERA Systems will present MERA RTU (MERA Retail and Transit Unit), a brand new softswitch designed to meet the needs of both wholesale and retail operators. Being a bundled product that comprises features of class 4 and class 5 VoIP products, MERA RTU delivers a set of unique benefits for wholesale and retail businesses.
With the comprehensive MERA RTU capabilities, wholesale carriers will be able to add services for retail customers to their business portfolio. Local exchange carriers or mobile carriers can increase the efficiency of the A-Z traffic routing originated from their retail customers. Large corporations will be enabled to build corporate voice network on a single, reliable multi protocol platform and to efficiently manage outgoing long distance calls. Lastly, carriers with both retail and wholesale traffic will save on network infrastructure by utilizing the same switching and control engine for all types of traffic. “Understanding the competitive landscape relating to VoIP market, we’re committed to support the development of VoIP industry by providing products and solutions like MERA RTU, which will help operators to save their money and double their system capabilities,” said Konstantin Nikashov, CEO of MERA Systems, “We have applied our rich experience in VoIP softswitch development to create MERA RTU for operators worldwide and accelerate the maturity of VoIP industry. We are proud to announce the launch of MERA RTU at such an IP-focused event as VON Expo 2009" MERA Systems invites you to visit booth 1505 to learn more about MERA RTU and MERA’s updated VoIP softswitch portfolio as well as get more information about MERA Systems special offers for partners.
For more information visit www.mera-systems.com
out of
found this valuable.
Do you find this valuable?
|
|
|
|
MERA Systems To Present Solutions For Mobile Carriers at Asian Carrier
|
The SIP School ~ News
|
0 |
Sep 8 2009, 5:30 AM EDT by
|
|
|
Thread started: Sep 8 2009, 5:30 AM EDT
Watch
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
out of
found this valuable.
Do you find this valuable?
|
|
|
|
MERA Systems to Announce the Company Initiatives for 4Q 2009 at IT EXP
|
The SIP School ~ News
|
0 |
Aug 28 2009, 5:31 AM EDT by
|
|
|
Thread started: Aug 28 2009, 5:31 AM EDT
Watch
MERA Systems will announce the company milestones for the end of 2009 at IT EXPO West on September 1-3, 2009 at Los Angeles, California.
This fall MERA Systems is going to start several corporate initiatives that will determine the company’s strategy for the upcoming year. Particularly, it will offer an upgraded VoIP product portfolio enhanced with a new version of MVTS II that features advanced capabilities. The company plans to prolongate its MVTS I to MVTS Pro Transition Program providing MVTS I customers with a sufficient discount on MVTS Pro and easing the process of migration to the next-generation VoIP softswitch. Also the company will launch partnership programs with the industry-leading vendors aimed at providing telecommunication carriers with comprehensive VoIP solutions for complex TDM replacement. Lastly the pinnacle of MERA Systems recent achievements will be its brand new product that comprises class 4 and 5 features to meet the needs of both wholesale and retail operators.
MERA Systems invites you to drop by at booth #434 to learn more about its updated VoIP softswitch portfolio and get more information about MERA Systems special offers.
For additional information about MERA Systems, please visit our website at www.mera-systems.com or address your sales representative at sales@mera-systems.com
out of
found this valuable.
Do you find this valuable?
|