Average Ratings 0 Ratings
Average Ratings 0 Ratings
Description
A hosted phone system, also known as PBX (Private Branch Exchange), enables businesses of all sizes to utilize an advanced enterprise-level telephone system without the significant expense associated with physical telephone infrastructure. This system is fully managed and operated by GlobalPhone, your provider of Voice-over-IP services. Operating in the cloud grants you the flexibility to work from various locations, including the office, home, or on a mobile device, while still having complete access to the office phone system. You gain access to features typically reserved for Fortune 500 companies, such as auto-attendants, call centers, hunt groups, conferencing options, call recording, and voicemail services. Unlike traditional phone networks that rely on copper wires to connect to an in-house PBX, SIP trunking eliminates the need for these outdated copper connections by employing Voice over Internet Protocol (VoIP) technology to establish a virtual phone line or trunk, facilitating the transmission of voice traffic to your phone system. This modern approach not only reduces costs but also enhances the overall functionality and reliability of your business communications.
Description
The Ingate SIParator® serves as an effective and adaptable Enterprise Session Border Controller (E-SBC) designed for secure and interoperable SIP connectivity, enabling seamless integration of PBXs and Unified Communications (UC) systems with SIP trunking providers. By streamlining SIP trunking processes, the SIParator facilitates the connection of remote UC endpoints, the aggregation of SIP trunks, and the distribution of sessions across various sites and service delivery points. It is widely used for securing Real-Time communications, ensuring SIP interoperability, and providing extensive connectivity options. This robust solution is compatible with all existing network infrastructures and includes a standard SIP proxy along with a SIP registrar. Furthermore, it supports Network Address Translation (NAT) and Port Address Translation (PAT), as well as Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) for encrypting both SIP signaling and media. By addressing common security concerns related to enterprise VoIP, the SIParator ensures a safe and dependable communication environment for organizations. Its versatility and reliability make it a preferred choice for businesses looking to enhance their communication frameworks.
API Access
Has API
No
API Access
Has API
No
Integrations
AWS Marketplace
No
Amazon Web Services (AWS)
No
Google Cloud Platform
No
Microsoft Azure
No
Rackspace OpenStack
No
Integrations
AWS Marketplace
Yes
Amazon Web Services (AWS)
Yes
Google Cloud Platform
Yes
Microsoft Azure
Yes
Rackspace OpenStack
Yes
Pricing Details
$14.75 per month
Free Trial
No
Free Version
No
Pricing Details
No price information available.
Free Trial
No
Free Version
No
Deployment
Web-Based
Yes
On-Premises
No
iPhone App
No
iPad App
No
Android App
No
Windows
No
Mac
No
Linux
No
Chromebook
No
Deployment
Web-Based
Yes
On-Premises
No
iPhone App
No
iPad App
No
Android App
No
Windows
No
Mac
No
Linux
No
Chromebook
No
Customer Support
Business Hours
Yes
Live Rep (24/7)
No
Online Support
Yes
Customer Support
Business Hours
Yes
Live Rep (24/7)
No
Online Support
Yes
Types of Training
Training Docs
No
Webinars
No
Live Training (Online)
No
In Person
No
Types of Training
Training Docs
Yes
Webinars
No
Live Training (Online)
No
In Person
No
Vendor Details
Company Name
GlobalPhone
Founded
1995
Country
United States
Website
www.gphone.com
Vendor Details
Company Name
Ingate Systems
Founded
2001
Country
Sweden
Website
www.ingate.com/session-border-controller-sbc
Product Features
Business Phone Systems
Call Logging
No
Call Monitoring
No
Call Recording
No
Call Routing
No
Employee Directory
No
IVR / Voice Recognition
No
Mobile Access
No
Virtual Call Center
No
Web Conferencing
No
Business VoIP
Call Parking
No
Call Recording
No
Contact Management
No
Encryption
No
IVR / Voice Recognition
No
Ring Groups
No
SIP Trunking
No
Unified Communications
No
Voice Quality Enhancement
No
Cloud PBX
Call Recording
No
Call Routing
No
IVR / Voice Recognition
No
Ring Groups
No
SIP Trunking
No
Virtual Extensions
No
Web Conferencing
No
SIP Trunk
Automatic Fail-over Routing
No
Call Data Records
No
Caller ID
No
Disaster Recovery
No
International Numbers
No
Multiple DID Options
No
PSTN Forwarding
No
Toll-Free Numbers
No
e911
No
Product Features
Firewall
Alerts / Notifications
No
Application Visibility / Control
No
Automated Testing
No
Intrusion Prevention
No
LDAP Integration
No
Physical / Virtual Environment
No
Sandbox / Threat Simulation
No
Threat Identification
No