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What is Zentrunk?

Zentrunk provides a cloud-based SIP trunking service designed for worldwide connectivity, accommodating an unlimited number of concurrent calls with a flexible pay-as-you-go pricing system. You can integrate your VoIP configuration in just a few minutes due to our meticulously developed and optimized network infrastructure, which alleviates any hassle on your part. Our service features low Post Dial Delay by seamlessly routing calls through a single hop to local carriers, in addition to offering essential functionalities such as Caller Line Identification (CLI) and Dual-Tone Multi-Frequency (DTMF) signaling, all while avoiding the issues associated with out-of-region audio looping. With a robust infrastructure that covers multiple regions and includes at least three local carrier connections in several countries, Zentrunk guarantees a remarkable uptime of 99.95%. This level of reliability positions it as an excellent choice for enterprises aiming to improve their communication systems. Furthermore, Zentrunk's commitment to customer satisfaction ensures that businesses can trust in our service for their operational needs.

What is SmartSIP?

SmartSIP® is a specialized software solution intended for on-premise applications, providing DTMF suppression and masking capabilities that enable contact centers to securely gather sensitive cardholder information during live interactions. It allows customers to enter their private data using their touch-tone phones while still being in contact with a representative. By employing DTMF suppression and masking, the tones created by the customer are replaced with flat tones that cannot be reverted, ensuring confidentiality. The interface used by agents presents a masked version of the customer's input, effectively eliminating the need to verbally disclose sensitive information. As a result, complete call recordings and detailed screen captures can be preserved without compromising security. SmartSIP operates as a SIP proxy between the Session Border Controller (SBC or Media Gateway) and the voice platform, managing DTMF data while maintaining the integrity of the audio connection. This cutting-edge tool is particularly designed for VoIP communication within contact centers, making it vital for safeguarding data security throughout customer exchanges. Additionally, implementing SmartSIP can greatly improve adherence to various regulatory requirements concerning data protection, further reinforcing the importance of its use in the industry. Its ability to facilitate secure transactions while enhancing compliance makes SmartSIP an invaluable component for modern contact centers.

Media

Media

Integrations Supported

3CX
Asterisk
Avaya Experience Platform
Cisco CX Cloud
Make

Integrations Supported

3CX
Asterisk
Avaya Experience Platform
Cisco CX Cloud
Make

API Availability

Has API

API Availability

Has API

Pricing Information

Pricing not provided.
Free Trial Offered?
Free Version

Pricing Information

Pricing not provided.
Free Trial Offered?
Free Version

Supported Platforms

SaaS
Android
iPhone
iPad
Windows
Mac
On-Prem
Chromebook
Linux

Supported Platforms

SaaS
Android
iPhone
iPad
Windows
Mac
On-Prem
Chromebook
Linux

Customer Service / Support

Standard Support
24 Hour Support
Web-Based Support

Customer Service / Support

Standard Support
24 Hour Support
Web-Based Support

Training Options

Documentation Hub
Webinars
Online Training
On-Site Training

Training Options

Documentation Hub
Webinars
Online Training
On-Site Training

Company Facts

Organization Name

Plivo

Date Founded

2011

Company Location

United States

Company Website

www.plivo.com/sip-trunking/

Company Facts

Organization Name

IntraNext Systems

Company Location

United States

Company Website

intranext.com/solutions/smartsip

Categories and Features

Business VoIP

Call Parking
Call Recording
Contact Management
Encryption
IVR / Voice Recognition
Ring Groups
SIP Trunking
Unified Communications
Voice Quality Enhancement

Cloud PBX

Call Recording
Call Routing
IVR / Voice Recognition
Ring Groups
SIP Trunking
Virtual Extensions
Web Conferencing

SIP Trunk

Automatic Fail-over Routing
Call Data Records
Caller ID
Disaster Recovery
International Numbers
Multiple DID Options
PSTN Forwarding
Toll-Free Numbers
e911

Categories and Features

Telephony

Auto-Dialer
Call Center Management
Call Monitoring
Contact Management
IVR / Voice Recognition
Inbound Reporting
Outbound Reporting
Predictive Dialer
Telemarketing Management
VoIP
Voice & Data Integration

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