Sign in or 
The SIP School ~ Wiki and Directory Discussion Forum
Categories : General Discussion
General discussion around all things related to this site.
General Discussion
General discussion around all things related to this site.
Share this
Discussion Spotlight
|
|
| Started By | Thread Subject | Replies | Last Post | ||||
|---|---|---|---|---|---|---|---|
| theblarney | Ingate Repacement | 2 | Sunday, 4:54 PM EST by theblarney | ||||
|
Thread started: Dec 9 2009, 6:23 PM EST
Watch
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
Show Last Reply
|
||||||
| ashishdubey1981 | NGTLive voip solution provider | 1 | Dec 14 2009, 2:01 AM EST by vocale | ||||
|
Thread started: Dec 12 2009, 7:59 AM EST
Watch
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
Show Last Reply
|
||||||
| dgolyanina | MERA Systems To Present Solutions For Mobile Carriers at Asian Carrier | 1 | Dec 10 2009, 3:18 AM EST by amir90p | ||||
|
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?
Keyword tags:
class 4/5
MERA
mobile solutions
voip softswitch
voip solutions
Show Last Reply
|
|||||||
| rouldph | Why is It This Way | 2 | Dec 10 2009, 3:16 AM EST by amir90p | ||||
|
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 ?
Show Last Reply
|
|||||||
| amir1983 | SIP server and bandwidth | 3 | Dec 10 2009, 3:12 AM EST by amir90p | ||||
|
Thread started: Dec 3 2009, 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?
Keyword tags:
sip server bandwidth
Show Last Reply
|
|||||||
| dry_aquaman | Cisco 7960 to Mitel 3300 | 4 | Dec 8 2009, 7:56 AM EST by dry_aquaman | ||||
|
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
Show Last Reply
|
|||||||
| ashishdubey1981 | NGTLive class 5 switching platform | 0 | Dec 1 2009, 7:51 AM EST by ashishdubey1981 | ||||
|
|
Thread started: Dec 1 2009, 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?
Keyword tags:
callback
calling card
class 5 switch
retail voip
|
||||||
| ashishdubey1981 | NGTLive encryption pc-phone and mobile dialer solution | 0 | Dec 1 2009, 7:49 AM EST by ashishdubey1981 | ||||
|
|
Thread started: Dec 1 2009, 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?
Keyword tags:
SIP Phones
|
||||||
| ashishdubey1981 | NGTLive class 5 switching platform | 0 | Dec 1 2009, 7:48 AM EST by ashishdubey1981 | ||||
|
|
Thread started: Dec 1 2009, 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?
Keyword tags:
class 5 switch
retail voip
|
||||||
| ashishdubey1981 | NGTLive Mobile dialer with encryption integrated | 0 | Dec 1 2009, 7:46 AM EST by ashishdubey1981 | ||||
|
|
Thread started: Dec 1 2009, 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?
Keyword tags:
mobile dialer
mobile voip
|
||||||
| Vijaysharma1 | Predictive Dialer,Call center SOftware,IVR | 0 | Nov 25 2009, 2:45 AM EST by Vijaysharma1 | ||||
|
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?
Keyword tags:
Call center SOftware
IVR
Predictive Dialer
|
|||||||
| sevanaoy | AQuA - Audio Quality Monitoring for Linux IP PBXs | 0 | Nov 24 2009, 8:31 AM EST by sevanaoy | ||||
|
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 |
||||||
| sevanaoy | AQuA - Audio Quality Analyzer for IP PBXs | 0 | Nov 24 2009, 8:28 AM EST by sevanaoy | ||||
|
|
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?
Keyword tags:
SIP Software
|
||||||
| JillRH | Broadvox SIP Trunking | 0 | Nov 18 2009, 2:10 PM EST by JillRH | ||||
|
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?
Keyword tags:
ASP
IP Network
ISP
ITSP
SIP Origination Termination
SIP Trunk
SMB
VoIP
|
|||||||
| ipcomms | Free SIP Trunking From IPComms- Now even Better! | 0 | Nov 18 2009, 1:09 PM EST by ipcomms | ||||
|
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 |
||||||
| JillRH | Broadvox Wins Award for GO! SIP Trunking | 0 | Nov 17 2009, 1:34 PM EST by JillRH | ||||
|
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?
Keyword tags:
carrier
ITSP
origination
SIP Trunk
SMB
telecommunications
termination
VoIP
|
|||||||
| Armi_Speedflow | Seeking for Egypt mobile, Germany mobile, Haiti mobile routes | 0 | Oct 14 2009, 10:44 AM EDT by Armi_Speedflow | ||||
|
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 |
|||||||
| pwiebe | Can't Find it | 0 | Oct 14 2009, 4:10 AM EDT by pwiebe | ||||
|
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?
Keyword tags:
aggregation
mysipswitch
open source
pbx
sip
|
|||||||
| hankkarl | ISDN, R2, RBS to SIP interworking module for IP PBXes | 4 | Oct 5 2009, 10:14 AM EDT by hankkarl | ||||
|
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.
Show Last Reply
|
|||||||
| grnvoip | grnVoIP provides quality wholesale voip termination | 1 | Sep 27 2009, 6:22 PM EDT by GaryShort | ||||
|
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?
Keyword tags:
sip termination
wholesale voip
Show Last Reply
|
|||||||
