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
Termination / Origination
Our software is support GSM calls termination , GSM calls origination , DID Numbers , SIP to Telegram calls termination , and SIP to whatsapp calls termination
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
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 and receive 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
You can easily monitor your calls traffic and route status through software main screen directly or through our website in real time
You can see call statuses (idle , dialing , connected ) for each channel , remaining minutes , idle time , 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
