PBXs - VoIP.ms Wiki

PBXs

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==FreePBX / Trixbox / 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://www.facebook.com/notes/voipms/back-to-basics-what-is-a-pbx/3004711782934047/ Back to Basics - What is a PBX?]
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[[File:Pbx-left.jpg|left]]  
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{|style="border-collapse: collapse; border:1px solid #000; width:500px"
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== 3CX Phone System ==
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|
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-
canreinvite=nonat<br>
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-
<nowiki>; nat=yes ; uncomment if behind nat</nowiki><br>
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-
context=from-trunk<br>
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-
host=atlanta.voip.ms<br>
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-
secret=johnspassword ;your password<br>
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-
type=friend<br>
+
-
username=100000 ;your account<br>
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-
disallow=all<br>
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-
allow=ulaw<br>
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<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki><br>
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fromuser=100000 ;your account<br>
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trustrpid=yes<br>
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-
sendrpid=yes<br>
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insecure=port,invite<br>
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-
qualify=yes<br>
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|}
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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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Register String
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[[3CX Phone System|3CX Phone System Configuration]]
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{|style="border-collapse: collapse; border:1px solid #000; width:500px"
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|
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100000:johnspassword@atlanta.voip.ms:5060
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|}
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<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>
 
-
==FreePBX / Trixbox / PBX in a Flash (IAX2)==
 
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https://www.voip.ms/m/samples/images/freepbxiax.gif
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== Asterisk ==
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[[File:Pbx-left.jpg|left]]
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[[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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{|style="border-collapse: collapse; border:1px solid #000; width:500px"
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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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|
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type=friend<br>
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username=100000 ;your account<br>
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secret=johnspassword ;your password<br>
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context=from-trunk<br>
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host=atlanta.voip.ms<br>
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disallow=all<br>
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allow=ulaw<br>
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insecure=port,invite<br>
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requirecalltoken=no<br>
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qualify=yes<br>
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|}
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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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Register String
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There are two standard methods to connect an Asterisk box to voip.ms:
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{|style="border-collapse: collapse; border:1px solid #000; width:500px"
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* [[Asterisk IAX2 | Asterisk (IAX2)]], to use the Inter-Asterisk protocol
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|
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* [[Asterisk SIP | Asterisk (SIP)]], to use the same standard Session Initiation Protocol used to connect to SIP phones
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100000:johnspassword@atlanta.voip.ms:4569
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* [[Asterisk PJSIP | Asterisk (PJSIP)]], to use the Open Source Embedded SIP protocol stack
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|}
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 +
 
 +
 
 +
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
 +
 
 +
[[Category: PBXes]]
<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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==Asterisk (SIP)==
 
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===sip.conf===
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== Avaya IP office ==
 +
[[File:Avaya-logo.png|300px|x200px|thumb|left|Avaya]]
 +
<br><br>
 +
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]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br/>
-
<nowiki>
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== Cisco IOS ==
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[general]              
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[[File:Cisco-new-logo-should-be.gif‎‎‎|300px|x200px|thumb|left|Cisco IOS]]
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register => 100000:johnspassword@atlanta.voip.ms:5060
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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.
 +
[[Cisco IOS|Cisco IOS Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
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<br>
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[voipms]
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== Elastix==
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canreinvite=no
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context=mycontext
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host=atlanta.voip.ms
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secret=johnspassword ;your password
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type=friend
+
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username=100000 ;your account
+
-
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 ;your account
+
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trustrpid=yes
+
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sendrpid=yes
+
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insecure=port,invite
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; nat=yes ; Uncomment this if your box is behind a NAT
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</nowiki>
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 +
[[File:Elastix_Logo.jpg‎|300px|x200px|thumb|left|Elastix]]
 +
<br><br>
 +
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|Elastix Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br><br>
-
===extensions.conf===
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== E-MetroTel==
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<nowiki>
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-
[mycontext]
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-
include => voipms-outbound
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include => voipms-inbound
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[voipms-outbound]
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[[File:E-MetroTel.png‎|300px|thumb|left|E-MetroTel - Exceptional Innovation]]
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exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
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<br><br>
-
exten => _1NXXNXXXXXX,n,Hangup()
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Designed for SMBs, Yeastar S-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing a solid, reliable and affordable on-premises and hosted business voice solution.
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exten => _NXXNXXXXXX,1,Dial(SIP/1${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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[[E-MetroTel|E-MetroTel Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br>
 +
[[category:E-MetreoTel]]
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[voipms-inbound]
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== FreePBX / PBX in a Flash==
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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:FreePBX_Logo.jpg‎|300px|thumb|left|FreePBX]]
 +
<br><br>
 +
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]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br>
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===sip.conf===
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== FreeSwitch==
-
Note: You'll need to create a sub account to use IP Auth
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[[File:Fslogo.png‎‎|300px|thumb|left|FreeSwitch]]
 +
<br><br>
 +
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]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br>
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<nowiki>
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== Grandstream UCM 6200==
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[voipms]
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canreinvite=nonat
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context=mycontext
+
-
host=atlanta.voip.ms
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-
type=friend
+
-
disallow=all
+
-
allow=ulaw
+
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; allow=g729 ; uncomment if you support g729
+
-
insecure=port,invite
+
-
; nat=yes ; uncommment if behind a nat
+
-
</nowiki>
+
 +
[[File:Grandstream-Logo-2018.png|300px|thumb|left|Grandstream]]
 +
<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>
 +
<br/>
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===extensions.conf===
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== Nortel/Avaya BCM 450 and BCM50 R6==
-
<nowiki>
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[[File:Avaya.jpg|300px|thumb|left|Avaya BCM]]
-
[mycontext]
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<br><br>
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include => voipms-outbound
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Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.
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include => voipms-inbound
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[voipms-outbound]
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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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exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
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[[NortelBCM|BCM Configuration]]
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exten => _1NXXNXXXXXX,n,Hangup()
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<div style="width:100%;overflow:hidden;clear:both"></div>
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exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
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<br><br>
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exten => _011.,n,Hangup()
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[[category:PBXes]]
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exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
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<div style="width:100%;overflow:hidden;clear:both"></div>
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exten => _00.,n,Hangup()
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<br/>
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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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<br><br>
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</nowiki>
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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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<br><br><br><br><br>
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==Asterisk (IAX2)==
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[[PBXes.org|PBXes Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br/>
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===iax.conf===
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== PhoneSuite Systems==
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<nowiki>
+
[[File:Phone-Suite-Logo-Color.png|300px|thumb|left|PhoneSuite Systems]]
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register => 100000:johnspassword@atlanta.voip.ms
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<br/>
 +
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.
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[voipms]
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[[PhoneSuite Systems|PhoneSuite Systems Configuration]]
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type=friend
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<div style="width:100%;overflow:hidden;clear:both"></div>
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username=100000 ;your account
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<br/>
-
secret=johnspassword ;your password
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[[category:PBXes]]
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context=mycontext
+
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host=atlanta.voip.ms
+
-
disallow=all
+
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allow=ulaw
+
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; allow=g729 ; uncomment if you support it
+
-
insecure=port,invite
+
-
requirecalltoken=no
+
-
</nowiki>
+
 +
== Positron Telecom Systems==
-
===extensions.conf===
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[[File:PositronLogo.jpeg‎|300px|thumb|left|Positron]]
 +
<br><br>
 +
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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<nowiki>
+
[[Positron_Telecom_Systems|Positron Configuration]]
-
[mycontext]
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<div style="width:100%;overflow:hidden;clear:both"></div>
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include => inbound
+
<br>
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include => outbound
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[[category:Positron]]
-
[outbound]
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== SIPfoundry==
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exten => _1NXXNXXXXXX,1,Dial(IAX2/109799@voipms/${EXTEN})
+
-
exten => _1NXXNXXXXXX,n,Hangup()
+
-
exten => _011.,1,Dial(IAX2/109799@voipms/${EXTEN})
+
-
exten => _011.,n,Hangup()
+
-
exten => _00.,1,Dial(IAX2/109799@voipms/${EXTEN})
+
-
exten => _00.,n,Hangup()
+
-
; inbound context example for your DID numbers, do not add the number 1 in front
+
[[File:Sipfoundry-logo.png‎|thumb|left|300px|Synway]]
 +
<br><br>
 +
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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[voipms-inbound]
+
To learn how to configure sipXecs to work with VoIP.ms, follow this 10-minute guide here:
-
exten => 7863643011,1,Answer() ;your DID
+
-
</nowiki>
+
 +
[http://myitdepartment.net/blog/191 Configuration]
-
==Cisco IOS==
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<div style="width:100%;overflow:hidden;clear:both"></div>
-
===SIP Trunk (Username/Password Authentication)===
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'''Known Issues'''
-
For the configuration below to work, you must have DNS name lookups properly configured on your router.
+
-
The example below is based on IOS 15.1(3)T.  Minor adjustments may be necessary for ealier IOS revisions.
+
-
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 #.
+
 +
RTP Time Out when a person leaves a Voicemail resulting in only a 60 second voicemail being possible.
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configure terminal
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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.
-
+
 
-
voice service voip
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   <action application="set" data="record_waste_resources=true"/>
-
   gcid
+
<div style="width:100%;overflow:hidden;clear:both"></div>
-
  clid substitute name
+
 
-
  allow-connections sip to sip
+
 
-
  no supplementary-service sip moved-temporarily
+
== Synway==
-
  no supplementary-service sip refer
+
 
-
  sip
+
[[File:Synway_logo.png|thumb|left|300px|Synway]]
-
  e911
+
<br />
-
  transport switch udp tcp
+
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.
-
  asserted-id ppi
+
 
-
  localhost dns:dns.name.of.your.device
+
[[Synway UC200| Synway UC200 configuration]]
-
  midcall-signaling passthru
+
<div style="width:100%;overflow:hidden;clear:both"></div>
-
  no call service stop
+
<br/>
-
+
 
-
sip-ua
+
== Talkswitch==
-
  credentials username your_account password 0 your_password realm voip.ms
+
 
-
  authentication username your_account password 0 your_password
+
[[File:TalkSwitch.png|300px|thumb|left|Talkswitch]]
-
  registrar 1 dns:newyork.voip.ms  !Pick your preferred first server
+
<br><br>
-
  registrar 2 dns:montreal.voip.ms !Pick the next best here
+
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.
-
  !You can configure up to 6 registrar servers for fault-tolerance
+
 
-
+
[[TalkSwitch|Talkswitch Configuration]]
-
!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
+
<div style="width:100%;overflow:hidden;clear:both"></div>
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dial-peer voice 1 voip
+
<br>
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  incoming called-number 5555551...
+
 
-
  voice-class sip asserted-id ppi
+
== Trixbox==
-
  no voice-class sip block 180
+
 
-
  no voice-class sip block 181
+
[[File:Trixbox_logo.jpg|300px|thumb|left|Trixbox]]
-
  no voice-class sip block 183
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<br><br>
-
  voice-class sip pass-thru headers unsupp
+
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).
-
  voice-class sip pass-thru content unsupp
+
'''Trixbox has not been maintained since June 2010. Customers should look for alternatives.'''
-
  voice-class sip pass-thru content sdp
+
[[Trixbox|Trixbox Configuration]]
-
  dtmf-relay rtp-nte
+
<div style="width:100%;overflow:hidden;clear:both"></div>
-
+
<br>
-
!This dial peer is for outgoing calls and matches anything.
+
 
-
!Finish dialing with a # to immediately route the call.
+
== Vodia PBX==
-
dial-peer voice 2 voip
+
 
-
  destination-pattern T
+
[[File:VodiaLogo.jpg|300px|thumb|left|Vodia]]
-
  voice-class sip asserted-id ppi
+
<br><br>
-
  no voice-class sip block 180
+
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.
-
  no voice-class sip block 181
+
[[Vodia PBX|Vodia Configuration]]
-
  no voice-class sip block 183
+
<div style="width:100%;overflow:hidden;clear:both"></div>
-
  voice-class sip pass-thru headers unsupp
+
<br>
-
  voice-class sip pass-thru content unsupp
+
 
-
  voice-class sip pass-thru content sdp
+
== Yeastar==
-
  dtmf-relay rtp-nte
+
 
-
  session protocol sipv2
+
[[File:Yeastar_Logo.png‎|300px|thumb|left|Yeastar]]
-
  session transport udp
+
<br><br>
-
  session target sip-server
+
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.
-
end
+
 
-
copy run start
+
[[Yeastar|Yeastar Configuration]]
 +
<div style="width:100%;overflow:hidden;clear:both"></div>
 +
<br>

Latest revision as of 03:32, 29 August 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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