Tekaba VoIP Gateway
Tekaba VoIP Gateway
Tekaba VoIP Gateway
Tekaba VoIP Gateway

Tekaba VoIP Gateway

0 Reviews

FREE

App specs:

License

PAID

Version

Tekaba VoIP Gateway 1.3.5.0

LatestUpdate

Last updated

OS

Windows 11
Windows Server 2022
Windows Server 2019
Windows Server 2016
Windows 10 32/64 bit
Windows Server 2012 R2
Windows Server 2012
Windows 2008 32/64 bit
Windows 8 32/64 bit
Windows 7 32/64 bit
Windows Vista 32/64 bit

Language

EN

As its name suggests, Tekaba VoIP Gateway is a SIP gateway application designed to convert telephony traffic into data that can be transmitted throughout a network. Based on the RFC 3261 protocol, it runs in the background as a Windows service, monitoring sent and received messages by a 3G USB Modem connected to your computer.

You start by plugging in the 3G modem to your PC and once the computer identifies the hardware device, you must configure the VoIP route. Tekaba VoIP Gateway displays a simple and intuitive interface that enables you to easily set all the requires parameters.

You can change the listening port (by default, this is 5080) and adjust the SRTP / RTP timeout. Tekaba VoIP Gateway supportsUPnP for NAT traversal and allows you to use an external IP address to forward incoming SMS messages as SIP requests. Such requests are then processed and the message content is sent back as a SMS.

Changing the SIP proxy is just as easy: you must enter the SIP domain, proxy address and used port, enter the username and password (if required) and change the registration timeout.

Before starting the service, Tekaba VoIP Gateway enables you to configure the telephony routes for VoIP to telephony message conversions. As such, you have to enter the source IP addresses and prefixes , destination prefixes and trunks. The application selects the best route it finds to send the SMS message. If the telephony routes are not configured, Tekaba VoIP Gateway will not be able to route VoIP calls and SIP requests.

Tekaba VoIP Gateway is compatible with the UDP, TCP and TLS transport protocols, also logging call transfers, thanks to the SIP REFER method. It enables you to keep an eye on all active calls and their status from the 'Ports' tab.

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