PBXs - VoIP.ms Wiki

PBXs

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==FreePBX / PBX in a Flash (SIP)==
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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.
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https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
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Take a peek at VoIP.ms Blog Article : [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
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[[File:PbxSIPtrunk.png]]
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== 3CX Phone System ==
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'''''Fill the blanks with your information, please note that the images above are just examples.'''''
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[[File:3CX Logo.jpg|300px|thumb|left|3CX Phone System]]
 +
<br><br>
 +
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.
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canreinvite=nonat
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[[3CX Phone System|3CX Phone System Configuration]]
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nat=yes
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<div style="width:100%;overflow:hidden;clear:both"></div>
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context=from-trunk
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host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
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secret=***** (password associated with the Main or Sub-account)
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type=peer
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username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
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disallow=all
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allow=ulaw
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; allow=g729 ; uncomment if you purchased g.729 from Digium
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fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
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trustrpid=yes
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sendrpid=yes
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insecure=invite
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qualify=yes
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Register String:
 
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youraccountnumber:yourpassword@atlanta.voip.ms:5060
 
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(i.e. 123456:mypass@atlanta.voip.ms:5060)
 
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Please note: Some customers have needed to change the fromuser 6 digit account number to their DID number for outgoing CallerID to be displayed as desired.
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== Asterisk ==
 +
[[File:Asterisk.png‎‎|300px|x200px|thumb|left|Asterisk]]
 +
<br><br>
 +
'''Asterisk''' is a telephone [[PBXs|private branch exchange]] (PBX), created in 1999 as open software for Linux and other UNIX-like systems.
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==FreePBX / PBX in a Flash (IAX2)==
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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 [[Order a DID Number|obtain a direct inward dialable number]] to receive calls and a trunk for outbound calls.
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https://www.voip.ms/m/samples/images/freepbxiax.gif
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Asterisk is at the heart of various products, such as [[FreePBX / PBX in a Flash (IAX2)|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.
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[[File:freepbxIAXtrunk.png]]
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There are two standard methods to connect an Asterisk box to voip.ms:
 +
* [[Asterisk IAX2 | Asterisk (IAX2)]], to use the Inter-Asterisk protocol
 +
* [[Asterisk SIP | Asterisk (SIP)]], to use the same standard Session Initiation Protocol used to connect to SIP phones
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* [[Asterisk PJSIP | Asterisk (PJSIP)]], to use the Open Source Embedded SIP protocol stack
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'''''Fill the blanks with your information, please note that the images above are just examples.'''''
 
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type=friend
 
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username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
 
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secret=***** (password associated with the Main or Sub-account)
 
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context=from-trunk
 
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host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
 
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disallow=all
 
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allow=ulaw
 
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insecure=port,invite
 
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requirecalltoken=no
 
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qualify=yes
 
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Register String:
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Asterisk is complex but powerful; complete information on its deployment and use would fill a book. See:
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youraccountnumber:yourpassword@atlanta.voip.ms:4569
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* 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
 +
[[Category: PBXes]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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'''NOTE''': The trunk name should be set to '''''voipms''''' in lowercase. Otherwise you may have issues with the incoming calls.
 
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If the trunk name is not specifically set to '''''voipms''''', the following error may result on inbound calls: "Call rejected, CallToken Support required."
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== Avaya IP office ==
 +
[[File:Avaya-logo.png|300px|x200px|thumb|left|Avaya]]
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<br><br>
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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.<br/>
 +
[[Avaya IP office|Avaya IP office Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br/>
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==Asterisk (SIP)==
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== Cisco IOS ==
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[[File:Cisco-new-logo-should-be.gif‎‎‎|300px|x200px|thumb|left|Cisco IOS]]
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<br><br>
 +
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.
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[[Cisco IOS|Cisco IOS Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>
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===sip.conf===
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== Elastix==
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<nowiki>
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[[File:Elastix_Logo.jpg‎|300px|x200px|thumb|left|Elastix]]
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[general]              
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<br><br>
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register => 100000:johnspassword@atlanta.voip.ms:5060
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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.
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[[Elastix|Elastix Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br><br>
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[voipms]
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== E-MetroTel==
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canreinvite=no
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context=mycontext
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host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
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secret=johnspassword ;your password
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type=peer
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username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
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disallow=all
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allow=ulaw
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; allow=g729 ; Uncomment if you support G729
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fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
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trustrpid=yes
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sendrpid=yes
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insecure=invite
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nat=yes
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</nowiki>
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*Note: As of FreePBX/Asterisk version 2.9.0.7 if ;comments are placed after the config entries as above it may prevent outgoing call's with the end points getting "All circuits are busy".  Remove the ;comments and the trunk will send the calls with no errors.
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[[File:E-MetroTel.png‎|300px|thumb|left|E-MetroTel - Exceptional Innovation]]
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<br><br>
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Designed for SMBs, Yeastar S-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing a solid, reliable and affordable on-premises and hosted business voice solution.
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===extensions.conf===
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[[E-MetroTel|E-MetroTel Configuration]]
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<nowiki>
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<div style="width:100%;overflow:hidden;clear:both"></div>
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[mycontext]
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<br>
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; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
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[[category:E-MetreoTel]]
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include => voipms-inbound
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include => voipms-outbound
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[voipms-outbound]
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== FreePBX / PBX in a Flash==
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exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
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exten => _1NXXNXXXXXX,n,Hangup()
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exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
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exten => _NXXNXXXXXX,n,Hangup()
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exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
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exten => _011.,n,Hangup()
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exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
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exten => _00.,n,Hangup()
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; inbound context example for your DID numbers, do not add the number 1 in front
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[[File:FreePBX_Logo.jpg‎|300px|thumb|left|FreePBX]]
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<br><br>
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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|FreePBX / PBX in a Flash Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>
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[voipms-inbound]
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== FreeSwitch==
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exten => 7863643011,1,Answer() ;your DID
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</nowiki>
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<br><br><br><br><br>
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==Asterisk IP Auth. (SIP)==
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[[File:Fslogo.png‎‎|300px|thumb|left|FreeSwitch]]
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<br><br>
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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|FreeSwitch Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>
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===sip.conf===
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== Grandstream UCM 6200==
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Note: You'll need to create a sub account to use IP Auth
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[[File:Grandstream-Logo-2018.png|300px|thumb|left|Grandstream]]
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<br/><br/>
 +
Grandstream Networks, Inc. has been connecting the world since 2002 with SIP Unified Communications solutions that allow businesses to be more productive than ever before. Our award-winning solutions serve the small and medium business and enterprises markets and have been recognized throughout the world for their quality, reliability and innovation. Grandstream solutions lower communication costs, increase security protection and enhance productivity. Our open standard SIP-based products offer broad interoperability throughout the industry, along with unrivaled features, flexibility and price competitiveness.
 +
<br/>
 +
[[Grandstream UCM6200 | UCM6200 Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
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<br/>
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<nowiki>
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== Nortel/Avaya BCM 450 and BCM50 R6==
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[voipms]
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canreinvite=nonat
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context=mycontext
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host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
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type=peer
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disallow=all
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allow=ulaw
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; allow=g729 ; uncomment if you support g729
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nat=yes
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</nowiki>
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 +
[[File:Avaya.jpg|300px|thumb|left|Avaya BCM]]
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<br><br>
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Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.
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===extensions.conf===
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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.   
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  <nowiki>
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[[NortelBCM|BCM Configuration]]
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[mycontext]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
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<br><br>
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include => voipms-inbound
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[[category:PBXes]]
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include => voipms-outbound
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br/>
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[voipms-outbound]
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exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
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exten => _1NXXNXXXXXX,n,Hangup()
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exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
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exten => _NXXNXXXXXX,n,Hangup()
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exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
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exten => _011.,n,Hangup()
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exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
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exten => _00.,n,Hangup()
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-
 
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; inbound context example for your DID numbers, do not add the number 1 in front
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-
 
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[voipms-inbound]
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exten => 7863643011,1,Answer() ;your DID
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</nowiki>
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<br><br><br><br><br>
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-
 
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==Asterisk (IAX2)==
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===iax.conf===
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<nowiki>
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register => 100000:johnspassword@atlanta.voip.ms
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-
 
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[voipms]
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type=friend
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username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
+
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secret=johnspassword ;your password
+
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context=mycontext
+
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host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
+
-
disallow=all
+
-
allow=ulaw
+
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; allow=g729 ; uncomment if you support it
+
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insecure=port,invite
+
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requirecalltoken=no
+
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</nowiki>
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-
 
+
-
 
+
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===extensions.conf===
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<nowiki>
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[mycontext]
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; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
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include => voipms-inbound
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include => voipms-outbound
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-
 
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[voipms-outbound]
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exten => _1NXXNXXXXXX,1,Dial(IAX2/voipms/${EXTEN})
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exten => _1NXXNXXXXXX,n,Hangup()
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exten => _NXXNXXXXXX,1,Dial(IAX2/voipms/1${EXTEN})
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exten => _NXXNXXXXXX,n,Hangup()
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exten => _011.,1,Dial(IAX2/voipms/${EXTEN})
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exten => _011.,n,Hangup()
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exten => _00.,1,Dial(IAX2/voipms/${EXTEN})
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exten => _00.,n,Hangup()
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; inbound context example for your DID numbers, do not add the number 1 in front
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== PBXes.org==
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[voipms-inbound]
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[[File:Pbxeshead.png‎|300px|thumb|left|PBXes]]
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exten => 7863643011,1,Answer() ;your DID
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</nowiki>
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<br><br><br><br><br>
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-
 
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==Cisco IOS==
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-
 
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===SIP Trunk (Username/Password Authentication)===
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For the configuration below to work, you must have DNS name lookups properly configured on your router.
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The example below is based on IOS 15.1(3)T.  Minor adjustments may be necessary for ealier IOS revisions.
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Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.
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-
This example uses newyork.voip.ms as a primary route and chicago.voip.ms as a backup route.  You can use whichever hosts you prefer, so long as you keep them consistent in the configuration.
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-
 
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configure terminal
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voice service voip
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  ip address trusted list
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  ipv4 74.63.41.218        !Current IP address for newyork.voip.ms at the time of this writing.
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  ipv4 173.208.83.50        !Current IP address for chicago.voip.ms at the time of this writing
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  ip address trusted call-block cause not-in-cug
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  gcid
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  clid substitute name
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  allow-connections sip to sip
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  no supplementary-service sip moved-temporarily
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  no supplementary-service sip refer
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  sip
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  e911
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  transport switch udp tcp
+
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  asserted-id ppi
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  localhost dns:dns.name.of.your.device
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  midcall-signaling passthru
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  no call service stop
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-
+
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sip-ua
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  credentials username your_account password 0 your_password realm voip.ms
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  authentication username your_account password 0 your_password realm voip.ms
+
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  registrar 1 dns:newyork.voip.ms auth-realm voip.ms  !Pick your preferred first server
+
-
  registrar 2 dns:chicago.voip.ms auth-realm voip.ms  !Pick the next best here
+
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  !You can configure up to 6 registrar servers for fault-tolerance
+
-
+
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!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to  (555)555-1999
+
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dial-peer voice 1 voip
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  incoming called-number 5555551...
+
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  no voice-class sip block 180
+
-
  no voice-class sip block 181
+
-
  no voice-class sip block 183
+
-
  dtmf-relay rtp-nte
+
-
  no vad
+
-
  voice-class sip bind media source-interface GigabitEthernet0/0      !Use your internet-facing interface here
+
-
  voice-class sip bind control source-interface GigabitEthernet0/0    !Use your internet-facing interface here
+
-
+
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!This dial peer is for outgoing calls and matches anything.
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!Finish dialing with a # to immediately route the call.
+
-
!This is the first-priority dial peer for if your first-priority registrar server is available
+
-
dial-peer voice 2 voip
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-
  destination-pattern T
+
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  no voice-class sip block 180
+
-
  no voice-class sip block 181
+
-
  no voice-class sip block 183
+
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  dtmf-relay rtp-nte
+
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  session protocol sipv2
+
-
  session transport udp
+
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  session target dns:newyork.voip.ms
+
-
  no vad
+
-
  codec g711ulaw
+
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  voice-class sip bind media source-interface GigabitEthernet0/0      !Use your internet-facing interface here
+
-
  voice-class sip bind control source-interface GigabitEthernet0/0    !Use your internet-facing interface here
+
-
  preference 1
+
-
+
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!This dial peer is the same as #2, but lower preference.
+
-
!This dial peer will be used for outgoing calls in case the first server is unavailable.
+
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!If ANY response (failure or success) is received from the first server, this dial peer will not be used.
+
-
dial-peer voice 3 voip
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  destination-pattern T
+
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  no voice-class sip block 180
+
-
  no voice-class sip block 181
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  no voice-class sip block 183
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  dtmf-relay rtp-nte
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  session protocol sipv2
+
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  session transport udp
+
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  session target dns:chicago.voip.ms
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-
  no vad
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  codec g711ulaw
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  voice-class sip bind media source-interface GigabitEthernet0/0      !Use your internet-facing interface here
+
-
  voice-class sip bind control source-interface GigabitEthernet0/0    !Use your internet-facing interface here
+
-
  preference 2
+
-
end
+
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copy run start
+
-
<br><br><br><br><br>
+
-
 
+
-
==Talkswitch==
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-
 
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-
[[File:TalkSwitch.png|300px|thumb|left|Talkswitch]]
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<br><br>
<br><br>
-
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.
+
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.
-
[[TalkSwitch|Talkswitch Configuration]]
+
[[PBXes.org|PBXes Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<div style="width:100%;overflow:hidden;clear:both"></div>
-
<br><br><br><br><br>
+
<br/>
-
==Trixbox==
+
== PhoneSuite Systems==
-
[[File:Trixbox_logo.jpg|300px|thumb|left|Trixbox]]
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[[File:Phone-Suite-Logo-Color.png|300px|thumb|left|PhoneSuite Systems]]
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<br><br>
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<br/>
-
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).  
+
PhoneSuite is a leading provider of hotel voice communication solutions for over 25 years it has leveraged its expertise in communication technology to provide high-quality, low-power consuming products. PhoneSuite is dedicated to the design and manufacture of products exclusively for the hotel industry.
-
'''Trixbox has not been maintained since June 2010. Customers should look for alternatives.'''
+
 
-
[[Trixbox|Trixbox Configuration]]
+
[[PhoneSuite Systems|PhoneSuite Systems Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<div style="width:100%;overflow:hidden;clear:both"></div>
-
<br><br><br><br><br>
+
<br/>
 +
[[category:PBXes]]
-
==3CX Phone System==
+
== Positron Telecom Systems==
-
[[File:3CX Logo.jpg|300px|thumb|left|3CX Phone System]]
+
[[File:PositronLogo.jpeg‎|300px|thumb|left|Positron]]
<br><br>
<br><br>
-
3CX Phone System for Windows is a software-based IP PBX that replaces a proprietary hardware PBX / PABX. 3CX’s IP PBX has been developed specifically for Microsoft Windows and is based on the SIP standard – making it easier to manage and allowing you to use any SIP phone (software or hardware).
+
Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. These features, which were traditionally only available to large companies allow you to be more efficient and productive.
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[[3CX Phone System|3CX Phone System Configuration]]
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[[Positron_Telecom_Systems|Positron Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<div style="width:100%;overflow:hidden;clear:both"></div>
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<br><br><br><br><br>
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<br>
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[[category:Positron]]
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==SIPfoundry==
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== SIPfoundry==
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[[File:Sipfoundry-logo.png‎]]
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[[File:Sipfoundry-logo.png‎|thumb|left|300px|Synway]]
<br><br>
<br><br>
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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.
+
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 a 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 making available our infrastructure to other interesting projects. A commercial enterprise edition of the sipXecs solution is available from eZuce with full support, software maintenance, training, indemnification, and other services.
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To learn how to configure sipXecs to work with voip.ms, follow this 10 minute guide here:  
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To learn how to configure sipXecs to work with VoIP.ms, follow this 10-minute guide here:  
[http://myitdepartment.net/blog/191 Configuration]
[http://myitdepartment.net/blog/191 Configuration]
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   <action application="set" data="record_waste_resources=true"/>
   <action application="set" data="record_waste_resources=true"/>
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<br><br><br><br><br>
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== Synway==
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==PBXes.org==
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[[File:Synway_logo.png|thumb|left|300px|Synway]]
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<br />
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Synway, as a world-leading VoIP enabling-technologies provider in China. has been specialized in providing superior Multimedia Gateway, Integrated Multimedia Switch, Telephony Hardware in use for Telecom communications.
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[[File:Pbxeshead.png‎|300px|thumb|left|PBXes]]
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[[Synway UC200| Synway UC200 configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br/>
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== Talkswitch==
 +
 
 +
[[File:TalkSwitch.png|300px|thumb|left|Talkswitch]]
<br><br>
<br><br>
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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.
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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.
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[[PBXes.org|PBXes Configuration]]
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[[TalkSwitch|Talkswitch Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<div style="width:100%;overflow:hidden;clear:both"></div>
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<br><br><br><br><br>
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<br>
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[[category:PBXes]]
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==Nortel/Avaya BCM 450 and BCM50 R6==
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== Trixbox==
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[[File:Avaya.jpg|300px|thumb|left|Avaya BCM]]
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[[File:Trixbox_logo.jpg|300px|thumb|left|Trixbox]]
<br><br>
<br><br>
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Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.
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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.'''
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[[Trixbox|Trixbox Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>
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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.
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== Vodia PBX==
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[[NortelBCM|BCM Configuration]]
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 +
[[File:VodiaLogo.jpg|300px|thumb|left|Vodia]]
 +
<br><br>
 +
Vodia PBX is a software-based, open standards IP PBX that offers complete UC functionality right out of the box. Vodia 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.
 +
[[Vodia PBX|Vodia Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br><br><br><br><br>
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<br>
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[[category:PBXes]]
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 +
== Yeastar==
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 +
[[File:Yeastar_Logo.png‎|300px|thumb|left|Yeastar]]
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<br><br>
 +
Designed for SMBs, Yeastar S-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing a solid, reliable and affordable on-premises and hosted business voice solution.
 +
 
 +
[[Yeastar|Yeastar Configuration]]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>

Latest revision as of 17:29, 12 November 2019

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.

Take a peek at VoIP.ms Blog Article : Back to Basics - What is a PBX?

Contents

3CX Phone System

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.

3CX Phone System Configuration


Asterisk

Asterisk



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 is complex but powerful; complete information on its deployment and use would fill a book. See:


Avaya IP office

Avaya



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

Cisco IOS



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

Elastix



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



E-MetroTel

E-MetroTel - Exceptional Innovation



Designed for SMBs, Yeastar S-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing a solid, reliable and affordable on-premises and hosted business voice solution.

E-MetroTel Configuration


FreePBX / PBX in a Flash

FreePBX



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

FreeSwitch



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


Grandstream UCM 6200

Grandstream



Grandstream Networks, Inc. has been connecting the world since 2002 with SIP Unified Communications solutions that allow businesses to be more productive than ever before. Our award-winning solutions serve the small and medium business and enterprises markets and have been recognized throughout the world for their quality, reliability and innovation. Grandstream solutions lower communication costs, increase security protection and enhance productivity. Our open standard SIP-based products offer broad interoperability throughout the industry, along with unrivaled features, flexibility and price competitiveness.
UCM6200 Configuration


Nortel/Avaya BCM 450 and BCM50 R6

Avaya BCM



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.org

PBXes



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.

PBXes Configuration


PhoneSuite Systems

PhoneSuite Systems


PhoneSuite is a leading provider of hotel voice communication solutions for over 25 years it has leveraged its expertise in communication technology to provide high-quality, low-power consuming products. PhoneSuite is dedicated to the design and manufacture of products exclusively for the hotel industry.

PhoneSuite Systems Configuration


Positron Telecom Systems

Positron



Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. These features, which were traditionally only available to large companies allow you to be more efficient and productive.

Positron Configuration


SIPfoundry

Synway



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 a 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 making available our infrastructure to other interesting 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:

Configuration

Known Issues

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"/>


Synway

Synway


Synway, as a world-leading VoIP enabling-technologies provider in China. has been specialized in providing superior Multimedia Gateway, Integrated Multimedia Switch, Telephony Hardware in use for Telecom communications.

Synway UC200 configuration


Talkswitch

Talkswitch



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.

Talkswitch Configuration


Trixbox

Trixbox



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


Vodia PBX

Vodia



Vodia PBX is a software-based, open standards IP PBX that offers complete UC functionality right out of the box. Vodia 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. Vodia Configuration


Yeastar

Yeastar



Designed for SMBs, Yeastar S-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing a solid, reliable and affordable on-premises and hosted business voice solution.

Yeastar Configuration


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