• Simple Environment

    PureVoIP SIP GSM gateway is works on Microsoft Windows operating systems , minimum recommended requirements Windows 10 English version 64x , I3 CPU , 4GB RAM , 5GB available free space

  • Easy to use

    PureVoIP SIP GSM gateway is very easy to use , any person has some basic skills on windows can use it , just connect your phones to our USB hub and run the software , that's it!
    All features can be managed through software GUI , no scripts/codes needed
    For better understanding please check our demo video

  • Multiple Routing Algorithms

    You can route calls based on channel ACD , ASR , channel load , or sequential

  • SIM block protection

    Our software has very important value which is prevent block of your used SIM cards in calls termination process , because you are operating the SIM cards inside normal phones (not unknown devices like normal GSM gateway devices) , this will make your SIM cards not easy to detect by your mobile operator , also we added in this solution many tricks to avoid SIM block :
    • Block guard module , when you enable this feature , the software will use a lot of tricks to avoid SIM block
    • Voice captcha to protect your SIM cards from generated calls by robots , [no FAS]
    • Manage delay between calls so the SIM activity will looks like human behavior
    • Manage calls routing to use lowest loaded SIM (Load balancing)
    • Manage black list/white list numbers to avoid suspicious numbers

  • Termination / Origination

    Our software is support both calls termination / calls origination

  • Callback Route

    When you or your agents are unavailable or busy , you can use our solution as answering machine to collect your clients messages as recorded audio files and contact them later

  • Channel Usage Control

    You can manage usage for each channel by set specific number of calls for each channel or specific number of minutes , after remaining minutes/calls attempts are finished , the software will eliminate this channel to avoid send IVR to caller

  • Dialing Plans

    You can receive different network traffic and assign each prefix to specific channels
    E.g , you can receive netwrok (A) traffic and terminate it using channels 1,2, and 3 , at the same time you can receive netwrok (B) traffic and terminate it using channels 4,5, and 6
    By default all channels will be available to terminate any incoming calls regardless the network

  • Support USSD and SMS

    You can check balance and recharge your SIM cards through our software , also you can send SMS

  • Whitelist and Blacklist

    If you have blacklist numbers you can load it to software to avoid dial these numbers , and you can also use the whitelist to pass calls only to the numbers on your whitelist.

  • SMPP Protocol Supported

    By using our solution , you can terminate SMS traffic with the following features:
    • Terminate SMS traffic side by side with calls traffic at the same time
    • Receive and terminate long text messages as concatenated segments
    • Keeping connection to SMPP server alive
    • Supported any language including Arabic, Chinese, Russian, and Greek
    • SSL/TLS support

  • Security

    You can secure your traffic by enable SIP and RTP encryption from software settings screen

  • Calls Recording

    You can activate calls recording , any call terminated / originated from software will be recorded on your PC (MP3) format

  • Static IP address is not needed

    No need for static IP , software need SIP account only to start receiving calls traffic
    You can connect your PC to internet to run the software through USB modem dongle , internet router , or any other way
    For better understanding please check our demo video

  • Bluetooth dongle is not needed

    No need for bluetooth dongles , our software connect to your phones through USB cables

  • Support all VoIP codecs

    All VoIP codecs are supported ( PCMU , GSM , G723 , PCMA , G722 , G728 , G729 , Speex , iLBC , L16 , G726 , OPUS ) , including bandwidth saver codec G729

  • Our package

    Our solution package including USB hub (device) multi channels + USB cables + PureVoIP software license + free subscriptions on SIPElectron softswitch

  • Custom issues

    No problems in customs , our hardware is USB hub only ( generic device / not categorized as VoIP device) , the shipment can pass any country in a normal way

  • Customization

    We are providing free software customization if is it simple and reasonable for our clients

  • NAT and firewall friendly

    PureVoIP SIP GSM gateway is friendly to those SIP clients who are behind the NAT/firewalls. If the software is behind the NAT/firewall, then it can easily be connected to your SIP server without making any extra settings (stun, estun, port forwarding).

  • Auto detect connected phones

    Our software will auto detect connected phones in 5 seconds , plug and play , without any configurations!
    For better understanding please check our demo video

  • Codec conversion - transcoding

    You can push your traffic in any VoIP codec , the software will convert it to the targeted codec

  • Compatible phones

    In general all old Nokia phones are compatible with our software , also Android phones are supported , you can check supported and tested phones on this page

  • Audio quality optimization

    We implemented new and powerful algorithm to reduce noise in calls and to enhance audio quality

  • Live monitoring

    Our software has nice interface for easy monitoring incoming calls , and you can see call statuses (idle , dialing , connected ) for each channel , and some statistics like ASR and ACD in general and for each channel , also you can get dialed numbers list for each SIM

  • Technical support

    Our technical support team will connect to your PC remotely to install , and run the software , also to provide you needed help / instructions about using the products with no problems and in best scenario

  • Pure VoIP offers affordable software customization for our products. Whether you need simple changes or need new functionality, we offer services to meet your needs.

    for more details , please contact development@pure-VoIP.com