From VoIP.ms Wiki
An acronym for Private Branch eXchange. PBX telephone systems support incoming calls from the outside PSTN, placing calls between users' phones (also known as extensions) and other phones or the outside PSTN, conferencing other users together, recording voicemails and a variety of other advanced telecommunication functions. PBX systems are broadly broken into several categories: traditional (also known as legacy); converged (also known as hybrid) or pure IP, aka IP-PBX. Traditional PBX systems usually either don't support IP at all or they support it only with expensive add-on equipment. Converged PBX systems support IP and PSTN connections with equal force. It is the most flexible and cost-effective model. IP-PBX systems, as the name implies, support only IP connectivity. Any PSTN connectivity must be achieved through external converters, known as Gateways.
3CX Phone System
3CX is a software-based, open standards IP PBX that offer complete Unified Communications, out of the box. 3CX makes installation, management, and maintenance of your PBX so easy that you can effortlessly manage it yourself, whether on an appliance, on your servers or in your cloud account.
Asterisk is a telephone private branch exchange (PBX), created in 1999 as open software for Linux and other UNIX-like systems.
Like other private branch exchanges, it allows attached telephones to make calls to one another and provides connections to outside lines to make and receive calls. To Asterisk, a VoIP provider represents a means to obtain a direct inward dialable number to receive calls and a trunk for outbound calls.
Asterisk is at the heart of various products, such as PBX in a Flash and Trixbox, intended to join multiple individual telephone extensions or devices as one office-style system. There are even versions of Asterisk which run under OpenWRT, an embedded Linux which was installable on some Linux-based Linksys routers.
There are two standard methods to connect an Asterisk box to voip.ms:
- Asterisk (IAX2), to use the Inter-Asterisk protocol
- Asterisk (SIP), to use the same standard Session Initiation Protocol used to connect to SIP phones
- Asterisk (PJSIP), to use the Open Source Embedded SIP protocol stack
Asterisk is complex but powerful; complete information on its deployment and use would fill a book. See:
- http://www.asteriskdocs.org is a free HTML book (the corresponding printed book is published conventionally by O'Reilly)
- http://www.asterisk.org is Asterisk's home site, operated by Digium.com
Avaya IP office
Avaya is a leading global provider of next-generation business collaboration and communications solutions, providing unified communications, real-time video collaboration, contact center, networking and related services to companies of all sizes around the world. IP Office is Avaya's telephone system for small and medium enterprises.
Avaya IP office Configuration
Cisco IOS (originally Internetwork Operating System) is software used on most Cisco Systems routers and current Cisco network switches. (Earlier switches ran CatOS.) IOS is a package of routing, switching, internetworking and telecommunications functions integrated into a multitasking operating system. Cisco IOS Configuration
Elastix is an open source unified communications server software that brings together: IP PBX, Email, IM and Faxing. The Elastix functionality is based on open source projects including Asterisk, HylaFAX, Openfire and Postfix. Those packages offer the PBX, fax, instant messaging and email functions, respectively, Elastix runs on CentOS operating system. Elastix Configuration
FreePBX / PBX in a Flash
As an open source, web-based PBX solution, FreePBX is easy to customize and adapt to your changing needs. FreePBX is currently being used to manage the business communications of all sizes and types of businesses from small one person SOHO businesses, to multi-location corporations and call centers. FreePBX / PBX in a Flash Configuration
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools. FreeSwitch Configuration
Nortel/Avaya BCM 450 and BCM50 R6
Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.
The BCM system is a popular legacy Nortel phone system that uses classic Nortel "Meridian" M and T series digital sets and their Unistim IP phones. BCM Configuration
PBXes is a hosted PBX running like a voicemail service or an email server in a data center. Like a classical PBX it serves as the bridge between your organisation (the extensions) and the public telephone network (the trunks). Thus it optimises both your internal and external communications. As trunks any SIP providers worldwide may be used.
SIPfoundry is the home of the sipXecs open source unified communications and collaboration project. If you are looking to replace your aging (IP) PBX with an all software modern communications solution that scales to mid-size and large enterprise, look no further. We are community dedicated to the development of SIP and XMPP based communications solutions. Founded in 2004 as an independent 501(3)c not-for-profit corporation based in Massachusetts, SIPfoundry is governed by an independent board of directors. SIPfoundry is open an invites all interested parties to cooperate and collaborate. While the sipXecs project is the largest active project at SIPfoundry, we are open to make available our infrastructure to other interested projects. A commercial enterprise edition of the sipXecs solution is available from eZuce with full support, software maintenance, training, indemnification, and other services.
To learn how to configure sipXecs to work with voip.ms, follow this 10 minute guide here:
RTP Time Out when a person leaves a Voicemail resulting in only a 60 second voicemail being possible.
Fix: Please add the following line to your record_waste_resources.xml or vars.xml file this will make RTP keep alive packets to continue to be sent while recordings are made.
<action application="set" data="record_waste_resources=true"/>
Unlike other small business telephone systems, TalkSwitch is easy to install, saving you time and money; you can even do it yourself. And with its easily configured settings, moving employees or changing the way your phone system handles calls is a snap.
Trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom). Trixbox has not been maintained since June 2010. Customers should look for alternatives. Trixbox Configuration