Author: VoIP Info
DID Service Providers in UAE
Telobal Provide UAE Toll-Free DID (Virtual Number) https://www.telobal.com/coverage/united-arab-emirates UAE Toll-Free DID ‘s available instantly on on-line store. Telobal.com also provide Free online pbx solutions (Inbound and outbound with your purchase number) Contact Email: info (a) telobal.com Contact Name:…
Read More »DID Service Providers in Pakistan
Tpad – The Global VoIP Network Official Website: https://www.tpad.com Service Name: Pakistan DID / Local Access Service (Free DID Service – covering 10 regions = Karachi, Gujarat, Faisalabad, Jhelum, Kharian, Lahore, Multan, Sargodha, Sialkot, Gujranwala) How it Works:…
Read More »DID Providers by Country
DID Service Providers in Bangladesh DID Service Providers in China DID Service Providers in Croatia DID Service Providers in Korea DID Service Providers in Pakistan DID Service Providers in Thailand DID Service Providers in UAE DID Service Providers…
Read More »Asterisk T.38
Be aware: T.38 is not T.38, there are still a great many interoperability issues out there! Version information Asterisk 1.2 has no support for T.38. Asterisk 1.4 supports only T.38 fax pass through; there is however a third…
Read More »caller id information for india
Hi, This process worked for me. using 1.TDM400P 4 port fxo card 2.asterisk-1.2.11.tar 3.zaptel-1.2.8.tar 4.libpri-1.2.3.tar 5.Bsnl phone line(pstn) i hope this will works for you. if still not, send me a email, if i get time, i will…
Read More »SineDialer ChangeLogs
For more information visit the SineDialer page. Version 3.0.2.0 • Major changes to the way communication is made with an Asterisk server. Now SineDialer has a funnel system which queues up information going to a server and then…
Read More »New Zealand Asterisk Details
Tips on Asterisk in New Zealand This guide offers some suggestions for getting Asterisk to work properly in New Zealand, specifically with a PRI purchased from TelstraClear connected to a TE110p card. For more tips on POTS lines,…
Read More »Asterisk Queue with limited calls per IAX agent
IAX2 does not have SIP-like call-limit configuration option. I was implementing a call queue, where clients used IAX2 softphone (idefisk). It was very disturbing for operators to have calls ringing on background of ongoing call, so I tried…
Read More »INOVE IP Telephony
Servicios de valor añadido para Telefonà a IP. Cisco IP Phone Services Los servicios avanzados para Telefonía IP permiten a las empresas aumentar la productividad de sus empleados, mejorar su imagen respecto a sus clientes y reducir el…
Read More »AstLinux Users Guide Chapter 2
20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ Page Contents Logging In Configuring your Keydisk Starting the Web Interface Logging In Access…
Read More »AstLinux Users Guide Chapter 1
Return to the AstLinux User Guide Contents 20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ Chapter 1 – Installation Page Contents Choosing…
Read More »AstLinux Users Guide Chapter 0
20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ About the Project Return to the AstLinux User Guide Contents Page Contents How it…
Read More »PJSUA
PJSUA is a command line SIP user agent (UA) written with PJSIP Open source SIP stack. While it is used mainly as the reference implementation of PJSIP, it is quite useful for testing or troubleshooting SIP installations, because…
Read More »PJMEDIA
PJMEDIA is an Open Source media stack written in C and is optimized for multimedia over IP communications. Among the features: it is optimized for small footprint, very very portable (Win32, Windows Mobile, Linux, *nix, MacOS X, Symbian,…
Read More »PJSIP
PJSIP is an Open Source Embedded SIP protocol stack written in C. The development of PJSIP is mainly focused on having a small footprint, modular, and very portable SIP stack for embedded development purposes (although it’s perfectly good…
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