Average Ratings 0 Ratings
Average Ratings 0 Ratings
Description
OpenSIPS is an open-source SIP proxy/server designed for handling voice, video, instant messaging, presence, and various other SIP extensions. This versatile signaling SIP server caters to carriers, telecom operators, and ITSPs, providing solutions for diverse applications such as Class 4 and Class 5 residential platforms, trunking and wholesale services, enterprise and virtual PBX systems, session border controllers, application servers, load balancers, IMS platforms, call centers, and much more. Its high throughput capability supports tens of thousands of calls per second and millions of simultaneous connections, making it a preferred choice for various SIP scenarios and services. OpenSIPS excels due to its flexible routing and integration options, which allow for custom routing logic through routing scripts and numerous interfacing APIs. With over 120 modules available, it facilitates effective application development for SIP handling, backend operations, and routing logic. Additionally, the 3.2 release of OpenSIPS underwent a thorough security audit, enhancing its safety and reliability for users across the board. This robust framework not only meets current demands but also continuously evolves to address future challenges in the telecommunications landscape.
Description
Wiretap Telecom provides an extensive range of VoIP and SIP trunking services designed for businesses in need of scalable, dependable, and secure communication options. Our intuitive user portal enables effortless management of services, featuring tools such as real-time analytics for calls and faxes, comprehensive trunk operation logs, and efficient call routing management. Additionally, it supports diverse messaging solutions like Email2SMS, SMS App, and smooth PBX integration, which caters to various messaging requirements. With the capability for international calls to any country and the convenience of managing multiple accounts through a single platform, Wiretap Telecom guarantees global connectivity and efficient operations. Our SIP trunking services work seamlessly with a variety of PBX systems, including FreePBX, Yeastar, 3CX, and Asterisk, making integration simple while enhancing voice functionality. By prioritizing user experience and operational efficiency, Wiretap Telecom stands out as a leader in innovative communication solutions for businesses.
API Access
Has API
Yes
API Access
Has API
Yes
Integrations
3CX
No
Ansible
Yes
Asterisk
No
Docker
Yes
Fonimo
Yes
FreePBX
No
Netdata
Yes
Yeastar Cloud PBX
No
Integrations
3CX
Yes
Ansible
No
Asterisk
Yes
Docker
No
Fonimo
No
FreePBX
Yes
Netdata
No
Yeastar Cloud PBX
Yes
Pricing Details
Free
Free Trial
No
Free Version
Yes
Pricing Details
No price information available.
Free Trial
Yes
Free Version
No
Deployment
Web-Based
No
On-Premises
Yes
iPhone App
No
iPad App
No
Android App
No
Windows
No
Mac
No
Linux
Yes
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
No
Live Rep (24/7)
No
Online Support
Yes
Customer Support
Business Hours
Yes
Live Rep (24/7)
Yes
Online Support
Yes
Types of Training
Training Docs
Yes
Webinars
Yes
Live Training (Online)
No
In Person
No
Types of Training
Training Docs
Yes
Webinars
No
Live Training (Online)
Yes
In Person
Yes
Vendor Details
Company Name
OpenSIPS
Country
United States
Website
opensips.org
Vendor Details
Company Name
Wiretap Telecom
Founded
2012
Country
United States
Website
wiretaptelecom.com
Product Features
Product Features
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