Dinstar VoIP GSM/WCDMA/LTE Gateways support to route VoIP calls to landline/mobile phone within the 2G/3G/4G mobile networks, provide a wide range of call cost-saving solutions for call centers, to increase answer rates from end users, and offer new effective operation methods for call centers.

Auto CLIP : Redirect call to original extension. DINSTAR GSM/LTE VoIP Gateway automatically stores information about outgoing calls to the Auto CLIP routing table. When a person calls back, the call will be routed directly to the original extension (e.g. receptionist) that made the former mentioned outgoing call.

SMS to Email : Allow users' email to receive the SMS of a GSM/LTE network. The SMS sent to the GSM/LTE ports will be received firstly by application of gateways and then forwarded to the pre-configured email address. Make users can receive the SMS through email.

Email to SMS : Detect users email address automatically. Identify the pre-configured content and forward to the assigned users number by SMS. Widely used for Alarm (Government), Notice (Education), Register& Tracking (Online shop, Logistics), Code/Receipt (Bank password) Auto Dialing / IVR : High voice quality, High AI recognition rate

AI Robot Interaction : Support mainstream Speech robot software, voice interaction via speech robot with artificial intelligence. Replace traditional phone seats, conversation with listeners in zero cost.

Call to click / Click to call : Allow provider have various ways to access customer services, such as Whatsapp, Facebook, Telephone, Email, Apps and Online consulting. Help call centers to serve customers anytime and anywhere, raise working efficiency and eventually improve customer satisfaction.

Softphone (SIP based) is no doubt the most important tool for your remote agents. Comparing with other ways, installing softphones on computers is easier, and technicians can help on this procedure via remote desktop tools. Prepare an installation guide for the remote agents and also some patience.

The desktop IP Phones can also be sent to agents’ locations, but make sure the configurations are already done on these phones as agents are not technical professionals. Now main SIP servers or IP PBXs support auto provisioning feature, which might make things easier than before.

These softphones or IP phones usually can be registered as remote SIP extensions to your main SIP server in headquarter of call center via VPN or DDNS (Dynamic Domain Name System). The agents can keep their original extensions and user habits. Meanwhile, a few settings need to be done on your firewall/router like port forwarding etc., which inevitably bring some security threats, an issue can’t be ignored.

To facilitate inbound remote soft phone and IP Phone access, Session border controller (SBC) is a key component of this system, be deployed at the edge of the call center network. When an SBC is deployed, all VoIP-related traffic (both signaling and media) can be routed from the softphones or IP phones over the public Internet to the SBC, which ensures all incoming / outgoing VoIP traffic is carefully controlled by the call center.

For outbound calls, simply use the agents’ mobile phones. Configure the agent’s mobile phone as an extension. When the agent makes outbound calls via softphone, SIP server will identify this is a mobile phone extension, and firstly initiate a call to the mobile phone number via VoIP media gateway connected to PSTN. After the agent’s mobile phone get through, SIP server then initiates the call to the customer. In this way, customer experience is the same. This solution needs double PSTN resources which outbound call centers usually have enough preparations.