Tekaba VoIP Gateway 1.3.2.0 Crack + License Key
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.
Download Tekaba VoIP Gateway Crack
| Software developer |
KaplanSoft
|
| Grade |
3.1
785
3.1
|
| Downloads count | 6026 |
| File size | < 1 MB |
| Systems | Windows Vista, Windows Vista 64 bit, Windows 7, Windows 7 64 bit, Windows 8, Windows 8 64 bit, Windows 2003, Windows 2008, Windows 2008 64 bit, Windows Server 2012, Windows 10, Windows 10 64 bit |
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 Serial 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 Serial 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.