Alternatives to OpenSIPS

3CX is an open-standards, software-based IP PBX that provides complete Unified Communications right out of the box. 3CX is suitable for all industries and sizes. It can provide everything you need, from mobility and status to advanced call center features and more, at a fraction the cost. 3CX is ideal for remote companies and includes mobile apps, video conference, and website live chat. The installation, management, and maintenance of the PBX are so simple that you can easily manage it yourself, on your own appliance, server, or on a cloud-based platform. You can also choose to have it hosted on 3CX for an easy solution.

PBXware is the first and most established IP PBX Professional Open Standards Turnkey Telephony Platform. PBXware has been providing flexible, reliable and scalable Next Generation Communication Systems (NGCS) and VoIP solutions to small and medium-sized businesses (SMBs), enterprises and Internet Telephony Service Providers, (ITSPs), call centers and governments around the world since 2004. This is done by combining the best of the latest technologies. Bicom Systems softswitch is available in the Business, Call Center, and Multi-Tenant Editions. Each edition supports specific features that maximize performance, reliability, expandability, and expandability.

Everything you need to know about the FreeSWITCH platform. FreeSWITCH, a Software Defined Telecom Stack, enables the digital transformation of proprietary telecom switches into a flexible software implementation that runs on all commodity hardware. FreeSWITCH is able to unlock the telecoms potential of any device, from a Raspberry Pi to a multi-core Server. FreeSWITCH, when combined with SignalWire, a hosted cloud platform, can connect with the outside world and scale up to any size. The pre-built virtual machine includes both a written as well as a video how-to. You only need a copy virtualbox to get started. You can also follow these instructions if you are a skilled developer to compile the FreeSWITCH platform directly from source. If you have a spare Debian instance, you can follow this link to find detailed instructions on how to install our packages.

Kamailio®, the successor to former OpenSER and SE, is an open-source SIP Server released under GPLv2+. It can handle thousands of call setups per seconds. Kamailio is a platform for VoIP, real-time communications and instant messaging. It can also be used to scale up SIP-to PBX gateways, PBX system, or media servers such as Asterisk™, FREESWITCH™, and SEMS. Among its many powerful features are asynchronous TCP and UDP, secure communication via VoIP (voice, text, and video); WebSocket support WebRTC; IPv4 or IPv6; simple instant messaging with embedded XCAP servers and MSRP relay; asynchronous operation; IMS extensions to VoLTE; ENUM, DID and least-cost routing; load balancing and routing failover; accounting, authentication, and authorization; support for many backend system such as MySQL, Postgres and Oracle, Redis, Cassandra and Memcached, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redisk, MongoDB and MongoDB, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redist, Redis, Redis, Redis, Casandra, Casandra, Redis, Casandra, MongoDB and Memcached, MongoDB, MongoDB, MongoDB, MongoDB, MongoDB, MongoDB, Casandra, MongoDB and Memcached, Casandra, MongoDB and Memcached

Signalmash is the ultimate Communication Platform as a Service solution. It offers a flexible suite of communication APIs that caters to the changing needs of businesses. Our offerings include: SMS API: Access enhanced customer interaction through a variety of text-based communication features, including alerts and authentication, promotional outreach, survey, appointment scheduling, as well as nurturing customer relationships. Flexible SIP Trunking - Simplify communication channels by using customizable SIP trunking. Businesses can benefit from customized contact center setups, personalized routing options, and encryption to ensure secure data transmission. VoIP PBX Solutions - Optimize your internal communications infrastructure with Signalmash VoIP PBX Solutions. These solutions streamline the setup and administration for private branch exchange systems and include intuitive call routing, voicemail, conference calling, and more.

Vodia is a software-based PBX that runs on Windows and Linux. It offers all the advanced features you would expect from a full-featured PBX, including call queues, auto attendants and call forwarding. Vodia is also the pioneer in Multi-Tenancy. You can host Vodia on a dedicated VM( Virtual Machine) in the cloud, premises, and maintain multiple FQDNs for each customer. Sign up for a hosted demo license code for 60 days.

*astTECS IPPBX system is the ideal solution for any business, large or small. *astTECS PBX has all the top-end features at an affordable price and is TEC certified by DoT (Govt of India). *astTECS IPPBX solution allows your business to be flexible, innovative, competitive, and future-ready while reducing telecom expenses. *astTECS Call Center Dialer is an omnichannel contact centre solution that offers advanced features and a real-time dashboard for agent monitoring. Advanced analytics and AI-based call center dialer will improve agent productivity and customer engagement. *astTECS provides a complete call recording solution for your company. *astTECS voice recorder is easy to use and offers advanced features. Integration with any PBX is seamless. Supports ISDN/PRI, IP, analog lines and mobile to record all incoming and outgoing calls.

It is easy to install and configure PBX software that is fully supported with our telephony hardware. It is ideal for small to medium-sized businesses (SMBs) who require a more sophisticated phone system than those found in larger companies. However, it does not come with a high price tag. OpenPBX combines the efficiency and ease of use of the desktop computer with the simplicity of a telephone handset to give you advanced features. You can also control your communication system from anywhere. Voicetronix offers line interface hardware that can connect to either standard POTS lines or primary rate (E1/T1) trunklines. OpenPBX's Do-it-Yourself (DIY), option allows customers to create their own phone system with commodity PC servers running Linux or analog telephone handsets. You can set up and manage your own phone system without the assistance of a technician. You can navigate to your web browser and select voicemail messages.

FusionPBX can be used to provide a highly-available single or domain-based multi-tenant VoIP, carrier-grade switch and call center server. It can also be used as a call center server. FreeSWITCH™, a multi-threaded, multiplatform communication platform, is highly scalable. It offers unlimited extensions, voicemail to email, music on hold and call parking. Call queues, phone provisioning, call center, call parking, and call center are all available. It offers the functionality you need and provides corporate-level phone system features for small, medium, and large businesses. It took a lot of time to fix bugs, tighten security, and improve the code. The future release is being worked on. We have made significant progress, and the master branch is now far ahead of the current release in many aspects. We are diligently working to complete the last changes required for this release.

HoduPBX is the best solution for business communication. HoduPBX's unique user interface and optimal user experience allow businesses to communicate efficiently with their end users. This software provides unparalleled connectivity in an office environment and remote locations, making communication seamless. HoduPBX empowers agents' work management with multitasking capabilities and efficiency. Our IP PBX software is a great choice for business communications. It supports 1000+ concurrent calls and has 10,000 extensions. HoduPBX IP PBX Software offers various sophisticated features such as Add-on Modules, Reseller Module, SIP Trunking, Queue Pro, Voice Transcription, and Webphone. It also offers white label features across IOS, Android, Chrome Extension, Windows, Mac, Linux, and ZohoPhonebridge Plugin.

Yeastar K2 IPPBX is available as both hardware and software. It supports 2000 users, 500 concurrent conversations, and 400 VoIP trunks across multiple locations. This gives medium and large enterprises full flexibility and capability. Yeastar K2IP PBX is designed to meet the communication needs of today and tomorrow. It offers enterprise-level features and exceptional reliability. It also allows remote working with next-level Linkus UC Softphone.

Asterisk is an open-source framework for building communication applications. Asterisk transforms an ordinary computer into a communications host. Asterisk powers IPPBX systems, VoIP gateways and conference servers, as well as other custom solutions. It can be used by small and large businesses, call centres, carriers, government agencies, and other agencies worldwide. Asterisk is open-source and free. Sangoma sponsors Asterisk. Today, more than 1 million Asterisk-based communication systems are in use in more than 170 nations. Nearly all Fortune 1000 customers use Asterisk. Asterisk is most commonly used by system integrators or developers. It can be used to create a complete business telephone system, enhance an existing system, or bridge between systems.

R.E. Communications offers an Asterisk-based IP-PBX system. These systems can support all industry-standard SIP phones and offer enterprise level features, which allow for up to 4000 phones. These systems also support SIP Trunking, which allows you to replicate the services of a Hosted solution while saving on per-seat costs by using SIP trunks and SIP phone. A typical 10 phone/seat system can support 4 to 6 SIP trunks, which is a significant savings over the 10 seats cost of a Hosted solution.

Taridium ipbx, a highly-capable software VoIP PBX system, replaces a traditional hardware-based PBX. ipbx can be run on standard server hardware or in the cloud. It does not require additional software licenses. An End User Panel allows you to manage all aspects of your extension. This includes listening to voicemail, setting up call forwarding, creating ad-hoc phone conferences, and more. You can set up your extension to forward only during certain times using your iCalendar feed, or based on a list favorite caller IDs. A conference room that is meet-me-type for all users. No more scheduling conflicts. Live conference management allows you to mute or kick participants. You can invite users by clicking a button, and the system will call them, instead of asking for cumbersome login instructions.

Avaya IP Office, a globally available and award-winning unified communications platform, is implemented and managed in your premises. Explore all the IP Office capabilities to improve your interactions with customers and employees. Set up call recording, tracking and reporting to study your customer calls. This Avaya call reporting capability can be delivered via IP Office to record and documentation incoming calls. You can also add dashboards and seats for real-time agents in call centers to improve call handling. Avaya IP Office and Avaya Spaces are combined to create a single app that includes voice, video, messaging and conferencing. This app is available on any device and in any location.

FreePBX stands as Freedom. FreePBX is the most popular open-source IP PBX in the world. It allows users to create a phone system that suits their needs. Open source communications is a powerful tool. Start building your phone system by taking a look at the many built-in features FreePBX offers. You can transform your FreePBX installation into a robust Unified Communications System with commercial modules from Sangoma or other add-ons. Visit the Knowledge Base at the FreePBX Wiki! FreePBX experts can help you with troubleshooting, installing, updating, and more. FreePBX is completely free to download and to use. Its power comes from a global network of developers who ensure that it remains a high-compatibility and customizable platform with all key features required to build a scalable business telephone system on any budget.

sipXcom was founded in January 2015 by the sipXecs development team at eZuce, Inc. sipXecs, SIPfoundry, and SIPfoundry had become a user community, and were not seeing any growth in their developer community. sipXcom has no restrictive contributor agreement. sipXcom source code licensing follows the copyleft-friendly AGLP version 3 (Affero General Public License). SWARM is the codename for the next generation sipXcom/sipXecs project. SWARM is currently in development and production readiness is expected for early 2017. It will be a microservices-based architecture which will improve sipX's scalability reliability, configurability, and configurationability. The software can be used with any compute platform, whether it is dedicated, virtual, cloud-based, or hybrid. You can also create hybrid instances that combine premise-based, data center, or cloud instances.

MikoPBX is a fully-featured phone system that offers high performance, stability, and rich functionality. MikoPBX can be installed on-premise (hardware install) or in any cloud service (Google Cloud, Amazon Cloud Solution and Microsoft Azure, for example). MikoPBX is free and provides basic business functionality. The system features a simple web interface that allows for call recording, conference calls and voice mail, voice mail pickup, call transfers and pick up, call queues, interactive voice response menus, and inbound and outside call rules. There are also holiday and night-time call switchers. MikoPBX can be extended with plugins and modules by developers from around the globe. MikoPBX is free and you can add-ons to enhance the capabilities of your PBX. MikoPBX does NOT license each user. You can expand your company without incurring additional costs for call recordings or new employees.

Enterprise interaction is a mashup between collaboration and communication. Secure enterprise interaction is possible from your private or hybrid cloud. Universal Cloud Communications Stack is a revolutionary concept that leverages big data architectures to enable real-time communications and collaboration. It allows for elastic scaling of transaction-oriented services at a global scale. The immediate benefits include rapid deployment and automated management that leverages standard infrastructure. Secure, scalable, and virtualized containers enable hosted and managed communications and collaboration services with ease. This allows MSPs to create new business models and allows enterprises to significantly lower their costs. Communications offer the greatest potential for cost savings across all IT. Combining modern software architecture and data-driven distributed infrastructure can transform an industry.

Yeastar P-Series PBX Systems is more than a PBX. It is a converged system designed for SMEs with higher expectations in business communications. The P-Series is a "PBX-Plus-More” solution that unifies voice, video, applications and collaboration. This allows SMEs to achieve more productivity, fill the UC gap and complete more complex tasks.

Dial-Office is an IP PBX software that offers flexible and feature-rich functionality for start-ups and SMBs who want to reap the benefits of VoIP technology. This IP PBX software solution is based on the SIP standard platform and offers advanced phone features that were only available with expensive circuit-switched enterprise telephone systems. Dial-Office IP PBX software provides reliable, low-cost unified communications for small- and medium-sized businesses using an established Local Area Network. This award-winning VoIP solution offers end-users crystal-clear voice quality, a variety of business management features, and the ability to lower communication costs and improve customer service. The following features are available on the platform:

VitalPBX is a unified communications PBX platform based on Asterisk (Centos 7). VitalPBX is a robust, scalable platform that will allow you to manage your PBX in a simple and intuitive manner. VitalPBX has a simple user interface and tooltips throughout the system. Telephony systems should be aware of security risks and take steps to avoid them. VitalPBX offers a variety of levels of protection against cyber threats in VoIP.

Digital transformation for collaboration and communications. AudioCodes Software-Defined Voice Network is an open solution that leverages the principles of Software-Defined Networking. It's designed to assist large enterprises in digital transformation. It simplifies the operation of their voice networks and improves call routing efficiency. This is achieved by consolidating communications silos, connectivity using SIP trunk services, and utilization of the corporate IP network for optimal call routing. AudioCodes' session boundary controllers (SBCs), global call routing and policy management tools, and voice network management tools allow SDvN to create a universal network that connects cloud-based and on-premises communications solutions.

Virtual IP PBX (Private Branch Exchange), or Phone System Appliance – CompletePBX is the heart and soul of Xorcom's hardware and software telephone systems. It's built on top of Asterisk and Linux platforms and is designed for stability and high performance. CompletePBX was built from the ground up to offer ease-of-use, advanced features, and security at all levels. Get a fully functional, free CompletePBX 5 virtual machine evaluation now! CompletePBX certification training is available online for free. It will guide you through the basics of the system. You can save a lot of time and effort if you do it yourself. We can help you get there quicker.

A2Billing, a class 4 and 5 softswitch, has inline billing and is designed to provide residential, business, and wholesale VoIP services. It also supports call-back, calling cards, and telephone number resale. Our professional support services are available. A2Billing, a free and open-source software for Asterisk, provides telecoms customer management, including agent, customer, and online signup pages. It also offers flexible inline billing and billing of calls and other services in realtime.

Wavetel was founded in 2010 and has since grown to be a major provider of telecom services in the UK, and around the globe. Wavetel began as a wholesale operator that offered Voice, SMS, and Data services to other operators. Wavetel's B2B services were launched in 2013 by the company. These include VoIP telephony services and Connectivity, Virtual Numbers. SIP Trunks. Microsoft Teams Direct Routing. Mobile, SMS, Call Center Solutions are available to its valued clients worldwide. Our voice messaging solutions have evolved over the years to be an enterprise solution. It can integrate with third-party CRMs and business apps through our programmable APIs.

thinQ provides a "better" way to power voice and SMS/MMS messaging applications in the cloud. Get up and running quickly with our integrated thinQ io portal and APIs. We make it easy to bring your 40 high-quality carriers to Twilio and Nexmo, Plivo and SignalWire, as well as enterprise contact centers and telecom VoIP providers. Our APIs allow you to quickly add and control inbound and outbound voice, texting, and other features to your existing tech stack. You can also connect to existing SIP providers to improve call delivery and quality. Our fully integrated thinQ.io Web platform makes it easy to purchase and port phone numbers, create route profiles, trunks, enable text messaging, as well as create route profiles, trunks, and other functions. You'll find a "better" way to communicate if you add call-specific analytics, SIP ladders, and our five-star support team to mitigate, route around outages, create customized routes and blocks, etc.

You can rely on our cutting-edge telecoms software and a powerful communications network to help your business do more. Our hosted solution will increase your outbound and internal dialing, and help you organize your calls while reducing costs. SIP Trunking for enterprises at wholesale rates. Unlimited capacity, high quality, high-volume VoIP services. Unique call routing, call recording, and reports will help you reduce office telephony expenses and increase functionality. Our network is purpose-built to handle high call volumes while ensuring exceptional call quality and reliability.

Access a wide variety of carrier services and SIP routing through an easy-to-use interface. This will allow you to offer more value for your customers or enterprise. VI Communication Services (formerly VoIP innovations) was created to enable Internet telephony service providers and managed service providers to provide high-quality services that customers expect. Our communication APIs can be used by businesses to enable their applications to use voice and messaging features to communicate with clients. Our CPaaS platform allows enterprises to seamlessly integrate communication capabilities in their software products using our prebuilt APIs over our robust SIP trunks. Our cloud-based communication APIs allow you to have more control and save money, while extending customer engagement and marketing across all communication channels like voice and messaging.

Unified Communications, OTT, and telecom providers can configure, launch, and monetize any type communication service in IP networks. This includes Unified Communications, Hosted IP PBX / IP Centrex and SIP Trunking. DID Management, DID Management, OTT Voice and calling cards. VoLTE/VoWiFI are some of the key functional modules. Key functional modules: Class 4 & 5 SIP softswitch, for residential and enterprise telephony service; convergent online billing, service provisioning and charging (optional).

Toll-free and local numbers are available from more than 160 countries. Forward incoming calls from any phone, PBX, or SIP. Our global network offers multiple layers of redundancy. You can make international outbound calls by using any of your business numbers as a Caller ID. Showing a local caller ID will increase your success rate. To make outbound calls, connect our SIP trunks with your phone system. You can easily set up and modify smart calling routing features such as auto-attendant /IVR, location and time-based routing, and simultaneous ringback tones. Custom greetings, call recording, voicemail, and call recording are all possible. You can easily add or remove virtual numbers, modify calling plans, download call detail records and invoices, modify advanced features, and reach technical support 24/7. Contact your account manager.

The D-TAC HD series billing system is a complete, post-paid CDR billing platform. It can be used with traditional switches, IP switches gateways, proxy servers and soft switches. This platform provides a quality-billing platform that can be used for high-density VoIP networks and TDM-based CDR networks. D-HD provides Telecom specific billing and business automation features that will make running your company easier. Every department in your company is given the screens, features and tools necessary to maximize productivity. The D-HD system has everything you need, from customer support systems to rate management tools. The D-TAC HD Billing System is the right choice for you if you need CDR management, cash receipts, carrier costing reporting, statistical route information, ASR reporting, powerful invoicing, cash receipts and powerful invoicing. Our system's core features include authentication and billing. We also have full AAA RADIUS capabilities and additional AARA.

Azure Route Server allows network appliances to dynamically exchange route information with Azure virtual Networks. Configure your network appliance and Azure ExpressRoute or VPN gateways so that they automatically receive the latest route information via Azure Route Server, instead of manually contacting each network. Azure Route Server can be used to enable dynamic routing instead of static routing between your gateways and network appliances in Azure. Azure Route Server exchanges routes using Border Gateway Protocol (BGP), a standard routing protocol. Azure Route Server can be used with existing deployments and new deployments. It supports any network topology such as hub-and-spoke, full mesh or flat virtual networks.

VoIP Route's hosted phone system uses the most recent cloud-based technology. VoIP Route connects your business without the high cost of traditional premise-based systems, and without any compromises. You can easily connect, call, and collaborate using your phone, smart device, computer, or laptop. VoIP Route is a business-class SIP trunk provider for both IP-PBX and analog/digital phone adapters or gateways. We have gained the trust of our customers and partners by providing the highest quality service at an affordable price, backed by outstanding support. Residential VoIP uses cutting-edge technology to deliver the best performance at a lower cost. Instead of using old phone lines to place calls you can convert your calls into digital data and use the Internet for your calls. VoIP residential phone service is cheaper than traditional phone service, and can help you save money on your monthly bills.

The Ingate SIParator®, a flexible, cost-effective and powerful Enterprise Session Border Controller, (E-SBC), is designed for SIP connectivity, security, interoperability, and interoperability. It connects PBXs and Unified Communications solutions to SIP trunking service provider providers. The SIParator simplifies SIP tunneling. It makes it simple to connect remote UC ends points, aggregate SIP trunks, and distribute sessions between sites or service delivery points. It is used for SIP interoperability, extensive connectivity, and real-time communications security. The SIParator®, which is compatible with all networks, comes with a standard SIP proxy as well as a SIP registry. It supports NAT and PAT as well as TLS and SRTP to encrypt SIP signaling as well as media. This eliminates the security problem that is often associated with enterprise VoIP.

Cool Telecom is a cloud-based, simple phone system that works for all sizes of businesses. We offer a multifunctional interface that includes CRM and Helpdesk tools. This allows sales teams to increase their sales and support teams to communicate clearly and effectively. The client can instantly add 50+ countries to their team, scale their teams based upon seasonality, and gain deep insights using real-time analytics. Accessible from any device. Place and take calls wherever you are. Cool Telecom multifunction app allows you to instantly get phone numbers in more than 50+ countries and make calls. You can quickly and efficiently route calls based on IVR selection, agent skills, and time zone. You can also get advanced analytics on agent and team performance, productivity, and monitor them on the live stream. Cross-reference data can be used with an existing CRM to gain more understanding and improvement.

JeraSoft Billing Platform, a multifunctional billing platform for mobile and business telecoms, such as business telephony and ISPs, CLECs and more, is available. JeraSoft Billing Platform allows users to automate the billing for various services. Jerasoft Billing Platform allows users to automate routing and rating processes, as well as mediation, while managing multiple subscribers.

Infinity Telecom is a Czech Republic licensed telecom operator. We engage in traffic exchange with partners around the globe (premium termination and wholesale termination). Infinity Telecom -- A Czech-based telecom operator group. We offer solutions in the areas data storage and transmission as well as IP and classic telephony, Internet access and content delivery, colocation, and other telecommunications services in many countries around the world. We offer solutions in these areas: - Global voice services (SIP trunking, DID numbers) - IP transit - Hosting services (server rental and colocation, hardware hosting. Content delivery, data storage, and similar services). Local services in Czechia & Slovakia (Internet, office telecom solutions). Our PoPs can be found in Prague, Bratislava Amsterdam, Amsterdam, Mumbai, Barranquilla and Mumbai.

Open-source control plane for service mesh. It provides security, observability and routing. Kuma, built on top of Envoy is a modern control plan for Microservices & Service Mesh, both for VMs and K8s. It supports multiple meshes within a cluster. The L4 + L7 policy architecture is out of the box to enable zero trust security and traffic reliability. Kuma is easy to set up and use. Kuma is natively embedded with Envoy proxy. It provides easy-to-use policies that can secure and observe, connect, route, and improve service connectivity for all applications and services, including databases. Modern service and application connectivity can be built across any platform, cloud, and architecture. Kuma supports Kubernetes environments, Virtual Machine workloads, and modern Kubernetes environments in the same cluster. Kuma also provides native multi-cloud connectivity and multi-cluster connectivity that can support the entire organization.

A Smart TelePhony Platform to Support Individual Business, Wholesale, and Residential VoIP Service Providers It is open-source. ASTPP is an open-source solution that allows anyone to start a telephony company without investing. ASTPP is a powerful VoIP Billing Software that provides a complete telephony solution. This open-source solution has gone beyond being a VoIP billing software to become "A Smart TelePhony platform". The latest version ASTPP has some new functional modules that are intended to eliminate the need for multiple solutions to run a VoIP company. It integrates hosted IP PBX, Class 4-/5 Softswitch, as well as complete invoicing and billing solutions that were developed using Smart Technology.

You can unblock websites you love and avoid government censorship. Proxynel proxy is the fastest web proxy browser that unblocks websites. By opening the app, you can browse and reach almost any blocked website within your country. The proxy browser app includes a feature that allows you to access US proxy servers. Protect your privacy online. This app is an online proxy browser that works with android. Your traffic is routed through US servers so there are no footprints. This makes it the ideal solution to avoid government surveillance. You can forget about the Google Captcha prompt when you search for something. Proxynel proxy app dynamically assigns a new IP to you every time you request it. This ensures that Google won't suspect anonymous traffic from the same IP, and prompts you to verify that you are not a robot. It's working!

Squid is a caching proxy that supports HTTP, HTTPS and FTP. It caches and reuses frequently requested web pages, which reduces bandwidth and improves response time. Squid is a powerful server accelerator with extensive access controls. It is compatible with all operating systems, including Windows. Squid is used worldwide by hundreds of Internet Providers to provide the best web access. Squid optimizes data flow between client/server to improve performance, cache frequently-used content to reduce bandwidth usage, and caches it to increase speed. Squid can route content requests to servers in many ways to optimize network throughput. Squid is used by thousands of websites all over the Internet to dramatically increase their content delivery. Squid can help reduce server load and increase delivery speeds to clients.

CenterPoint Securities offers a powerful range of services, products, and partnerships that will help you build and manage your trading company. CenterPoint Securities is a stock broker with direct market access that allows clients to control their order flow. DAS Trader Pro allows direct access trading and features advanced order types, charting and multi-account management in a live-time environment. Sterling Trader®, Pro is a front-end, direct access trading platform that can be used for multiple purposes. It has a number of unique features. CenterPoint is a competitive option for traders who rely heavily on shorting. We have done this by selecting clearing firms with excellent easy to borrow lists and industry-leading locate providers. Our route list is unrivaled. We are constantly improving and taking into consideration the suggestions of our clients to make new routes. The institutional routing options available to retail clients are extensive.

IBM Sterling Secure Proxy protects your trusted network by blocking connectivity between internal servers and external partners. It can protect your network and data exchanges at network edge to allow trusted business-to-business transactions as well as managed file transfer (MFT), file exchanges. Secure Proxy is a proxy that operates in a demilitarized area (DMZ). It uses multifactor authentication, SSL session break, closing of inbound firewall gaps, protocol inspection, and other controls to protect your trusted zone. Firewall navigation best practices are provided to enhance your data and trusted zones security. Multifactor authentication is provided before connecting to backend systems for tighter controls. Proxy support is available and functions are provided to increase the security of edge-based file transfers. Advanced routing allows for faster infrastructure changes and reduces risk during migrations.

Residential Proxies (also known as Static Residential Proxies), Datacenter Proxies and Mobile Proxies are all services that offer users a step up in security and speed. - Unlimited Bandwidth Proxies and Blazing Fast Proxies - High Level of Security - You can use the same IP address multiple times. - 99.98% Uptime plus 24/7 365 Support - Superior service for Account Management, Web Scraping and Social Media, SERP and other use cases

Mockoon has tons of features that make API mocking easy. Get the latest Mockoon version and start creating mock APIs. You can create unlimited mock APIs with unlimited routes and run them simultaneously. You can use the CLI to run mock APIs in any automated or headless environment, including servers, CI, and GitHub Actions. Swagger/OpenAPI format support for mock API import/export. Multiple rules-triggered responses or random responses can be served with any headers body or HTTP status codes. Your mock API can be served over TLS using a self-signed certificate. Add any response headers in your routes and mock API. Auto-completion. You can add latency at the environment, route, or both. For easier debugging, all incoming and outgoing requests are recorded. Mockoon's proxy mode allows you to redirect all routes not defined to the specified host. File serving with automatic mime type detection, templating support.

The Nokia Virtualized Service Router is a highly flexible virtualized edge router that can be used in telco cloud environments. It allows for rapid service innovation and lower operating costs. Service providers can increase their service reach, open new markets, or accelerate time to market with the VSR. The VSR supports a wide range of virtualized IP edge apps and virtualized network functions (VNFs), to support residential, business, and mobile services. The VSR is based on the Service Routing Operating System, (SR OS). It delivers high performance and elastic scalability in x86 server environments. The VSR provides the highest level of reliability and resilience. It also offers flexible management options - from working with open frameworks to delivering OpenStack®-integrated VNF management and element management capabilities with our 5620 Service Aware Manager (SAM).

NinjaProxy offers enterprise-class proxy IPs which are fast, reliable, and secure. We help businesses and brands in the search engine and internet advertising industries. We offer a variety of IP addresses including Residential Proxy Servers and 4G/5G Mobile Proxy Servers.

Go Trunk is a SIP-based trunking service that provides a cost-effective, reliable, and feature-rich communications platform for businesses with their own PBX. It connects your outgoing calls with the PSTN (public switched phone network) and provides inbound telephone numbers. Go Trunk offers powerful communication features for businesses, agility, reliability, and low-costs. The latest developments in SIP trunking will be covered in our blog posts. These articles will introduce the concept of SIP trunking and discuss the advantages of SIP trunking over traditional telecoms solutions like ISDN. These posts will help you understand how to set-up your SIP trunking, and the best ways to harness the power and capabilities unified communications to provide competitive advantage for your company. There is a lot to be learned about VoIP. However, most of it seems to be focused on money.

Reduce the cost of owning your business' comms infrastructure. Pay as you go. Global network. Direct routes. Volume and usage discounts. Our SIP and IAX solutions are top-notch. We offer features like elastic trunking, instant provisioning and direct interconnections. We can handle all your SIP enterprise requirements for call termination and origination, no matter how many agents you have. CALLR handles both decoding and encoding of your signal. All major codecs are supported, both in- and out. Our solution works with all major VoIP software, including Asterisk, Vicidial and Freeswitch. We can handle all your calling needs, outbound and inbound. Our smart routing system allows us to offer lower rates and reduce your call volume. Our SIP and IAX network can be accessed worldwide. We can provide reliable service with local interconnection via data centers wherever you are located.