Tekaba VoIP Gateway 1.2.0 Crack + [Mac/Win]
Tekaba VoIP Gateway Crack is a SIP Gateway that provides users with the advantages of a “Call Center” connected to their computer via a 3G modem. By running in the background as a Windows service, it converts telephony traffic into data that can be transmitted throughout a network.
This application converts such traffic with an adapter associated with a GSM (third generation) or CDMA (fourth generation) card. It supports conversion between and SIP-UDP, TCP and TLS protocols.
Tekaba VoIP Gateway Crack has a panel of configuration and monitoring parameters that include:
The software interface: it displays a very graphical interface, very easy to configure and with easy access to all required parameters
The configuration: is user-friendly, allowing you to enter all required parameters: each parameter has its own menu, allowing you to easily select it and modify it
Quality of service: Tekaba VoIP Gateway Activation Code can enable or block calls, enable or block media traffic, block and force calls on hold, block all inbound and outbound traffic.
The monitoring parameters: this tool allows you to monitor all the parameters that you have modified in the configuration panel, you can trace the path of the calls and SMS messages you have received through the events panel. The 3G modem can also be monitored by Tekaba VoIP Gateway Crack For Windows.
All of these parameters can be configured individually for each user, group of users or for all users. This allows you to achieve a “Call Center” service provisioned according to your needs.
Tekaba VoIP Gateway SIP Gateway:
Because SIP is a protocol used by VoIP telephony gateways, it is compatible with the SIP protocol. It supports RTP for streaming media, including conferencing and voice recording. It also provides quality of service (QoS) and monitoring by switching the parameters for each user individually. It can enable or block SMS messages, either inbound or outbound, or both at the same time.
Tekaba VoIP Gateway has a default location of Tokyo because some SIP endpoints prefer the local domain to the external domain. For this reason, Tekaba VoIP Gateway has the option to automatically select a local endpoint for the SIP traffic. You can also choose the resource location of the interface used by the gateways.
Tekaba VoIP Gateway supports all of the existing RFC 3261 (SIP protocol version 1) extensions. It has the support of UPnP for NAT travers
Tekaba VoIP Gateway 1.2.0 With Key Free Download For Windows
* Microsoft UPnP discovery mechanism
* Support for several protocols, including SIP, 3GPP, SS7, RFCOMM and SCCP
* VoIP routing
* Support for up to 10 concurrent VoIP calls
* TCP/UDP/TLS and SIP proxies
* Automatic detection of proxy address and port
* Support for proxying UDP and SCTP datagrams
* Support for calling-party restriction
* SIP messages logging (EXPIRES, BYE,...)
* Dialling registration
* Support for SCCP SCF
* GSM MSC redirects
* SMS forwarder
* Call transfer support
* Display caller ID and callee ID in the 'Port' tab
* CAMEL/CALLIDP support
* DVrP support
* Interactive call map
* Call forwarding support
* Call display in the 'Port' tab
* CLI support (Command-Line Interface)
* Web GUI: new GUI that is more integrated with the Windows Explorer and supports plug-ins (by SWREG)
Other features:
* SIP over HTTPS transport
* Active mode is enabled by default, this is very important for LTE operators as it maximizes the performance for calls made over LTE (unfortunately, this has an impact on latency)
* Uses a secure SIP Transport Layer Security (TLS) connection if available, otherwise falls back to the insecure clear text SIP protocol
* Uses the Secure Real Time Transport Protocol (SRTP) to encrypt the data
* Encrypts the phone book and user information
* Encrypts all calls
* Built-in dial plan for VoIP routing
* Number portability (lookup through GEO database with Sprint and Qwest)
* Call rate control
* Call Forwarding
* Call parking
* Call restriction
* Call control
* Call recording (with SIP Recording Agent)
* Call return to sender (SIP return-to)
* Call ID translation
* Call waiting (SIP wait-for-incoming)
* Contact rerouting (SIP contact-transfer)
* Fluency grade
* Fax (SIP fax-application)
* GPS reporting
* MMS
* Subscription (1 week)
* Text-to-voice (SIP Text-to-Speech)
* Real-time text display
* Trunking
* Anomaly
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