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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Joins Internet and local network/subnet in one network
Use a LAN IP address to forward calls
Sends outbound and inbound calls
On call arrival, build a SIP-enabled fire-wall
Redirects calls with REFER
Shows call statistics
Log transfers
Call home
IP-PBX Manager
IP PBX Manager is a free software program for Windows
(XP, Vista, 7,8) which allows you to create and use the service networks, IP telephones, contact lists and to manage the calls in them.
If your computer is connected to the Internet, you can use the program to manage your physical IP PBX telephone switch directly from your computer.
IP PBX Manager can be installed on your computer without any coding experience and provides all the features necessary for managing VoIP users.
IP PBX Manager is a very fast and easy to use software which allows you to connect all kinds of telephony hardware (internal and external) directly to your computer without any special cable.
Freecall
Freecall is a Windows application that aims to simplify the administration of VoIP solutions (such as analog telephone adapters, external servers, etc.) from a networked point of view. Freecall is a software package built in such a way as to provide a maximum of convenience and ease of use when it comes to setting up and configuring the various components needed for a properly running VoIP service.
Freecall is quite an advanced software, but it also comes with a lot of powerful settings that allow for a broad range of customization and a complete freedom to execute your own solutions.
Basically, the application allows you to add an external TCP / IP server as a SIP Call Agent and to configure its settings (it can run on a network, an embedded system, etc.)
Additionnally, Freecall includes all the necessary files for installing your external server and for configuring it for the various operations.
IRMngr
IRMngr is a utility that keeps track of which SIP messages have been processed and which SIP messages have been abandoned and which SIP messages have been forwarded for eventual retransmission. IRMngr is the most powerful VoIP monitoring software application available on the market.
IRMngr is simply the all-in-one solution to take care of all your VoIP monitoring needs. Providing both monitoring of SIP messages in progress
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* 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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Tekaba VoIP Gateway 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 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 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.
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
What’s New In Tekaba VoIP Gateway?
Vocus’ Unified Communications (UC) SIP Trunking Solution is a unified IP-PBX for SMB and Enterprises. It can be used in conjunction with SIP trunking to provide a cost-effective means of using SIP with Asterisk. UC SIP Trunking solution is a complete IP-PBX solution enabling SIP trunks to be created at any time without limitation or restriction. All pre-installed features include call recording, auto attendants, IVR and SoftPhone for i-mode and PC-Telephony. UC SIP Trunking can also be used with SIP-BRI/H323 and PRI trunks. The solution is 100% Asterisk-based and all the front-end and back-end voicemail can be integrated with the PBX. UC SIP Trunking is extremely powerful and fully customizable with scripting and APIs.
Important: To have a complete UC SIP Trunking solution you need UC SIP Trunking Server, UC SIP Trunking Plug-in and Asterisk.
3) UC SIP Trunking Server – Supports Asterisk for all SIP Trunks.
4) UC SIP Trunking Plug-in – Access to SIP Trunks.
5) Asterisk PBX – Supports Voicemail and Softphone.
Features:
* UNIX OS.
* Multiple channels and FXS/FXO as well as Datapath and Channel.
* SIP trunks creation at any time.
* Up to 4,000 SIP users for a single IP address.
* PBX management.
* Up to 1,000,000 calls per hour.
* Easy to use interfaces.
* Support for UAC (Unidentified Caller: Caller ID redirection).
* Built-in Real-time Recording (RTR) with JPEG/MPEG/DAT files support.
* Support for SIP outbound (from the directory) and SIP inbound (from outside in the network) registration.
* Screening and blocking for SIP callers that are not in the internal list.
* Call-back service to the directory (Reverse lookup).
* Call-back service to the outside caller, in association with the name of the caller.
* Called number forwarding.
* Multi-phone numbers (
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Additional Notes:
Recommended:
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RAM: 4 GB
HDD: 70 GB
GPU: 1 GB
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