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PBXs

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==FreePBX / PBX in a Flash (SIP)==
==FreePBX / PBX in a Flash (SIP)==
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https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
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[[File:FreePBX_Logo.jpg‎|300px|thumb|left|FreePBX]]
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<br><br>
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[[File:PbxSIPtrunk.png]]
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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.
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[[FreePBX / PBX in a Flash (SIP)|FreePBX / PBX in a Flash (SIP) Configuration]]
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'''''Fill the blanks with your information, please note that the images above are just examples.'''''
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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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canreinvite=nonat
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nat=yes
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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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==FreePBX / PBX in a Flash (IAX2)==
==FreePBX / PBX in a Flash (IAX2)==
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https://www.voip.ms/m/samples/images/freepbxiax.gif
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[[File:PIAF.gif‎‎|300px|thumb|left|PBX in a Flash]]
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<br><br>
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[[File:freepbxIAXtrunk.png]]
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If you've longed for the good ol' days of Asterisk@Home, welcome back to the new steroid-enhanced version. PBX in a Flash™ is the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users and VARs. You'll have a high-performance turnkey Asterisk PBX that's easy to upgrade with dozens of add on scripts to provide virtually any feature you can imagine.
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[[FreePBX / PBX in a Flash (IAX2)|FreePBX / PBX in a Flash (IAX2) Configuration]]
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'''''Fill the blanks with your information, please note that the images above are just examples.'''''
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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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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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youraccountnumber:yourpassword@atlanta.voip.ms:4569
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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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==Asterisk (SIP)==
==Asterisk (SIP)==
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[[File:TC-22622-MainIcon.gif‎‎|300px|thumb|left|Asterisk]]
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===sip.conf===
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<br><br>
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Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide.
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<nowiki>
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[[Asterisk (SIP)|Asterisk (SIP) Configuration]]
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[general]               
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<div style="width:100%;overflow:hidden;clear:both"></div>
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register => 100000:johnspassword@atlanta.voip.ms:5060
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[voipms]
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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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===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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[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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; inbound context example for your DID numbers, do not add the number 1 in front
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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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==Asterisk IP Auth. (SIP)==
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===sip.conf===
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Note: You'll need to create a sub account to use IP Auth
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<nowiki>
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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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===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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[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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; inbound context example for your DID numbers, do not add the number 1 in front
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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>
<br><br><br><br><br>
==Asterisk (IAX2)==
==Asterisk (IAX2)==
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[[File:TC-22622-MainIcon.gif‎‎|300px|thumb|left|Asterisk]]
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===iax.conf===
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<br><br>
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Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide.
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<nowiki>
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[[Asterisk (IAX2)|Asterisk (IAX2) Configuration]]
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register => 100000:johnspassword@atlanta.voip.ms
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<div style="width:100%;overflow:hidden;clear:both"></div>
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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)
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disallow=all
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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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===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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[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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[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>
<br><br><br><br><br>
==Cisco IOS==
==Cisco IOS==
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[[File:Cisco-new-logo-should-be.gif‎‎‎|300px|thumb|left|Cisco IOS]]
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===SIP Trunk (Username/Password Authentication)===
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<br><br>
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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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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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The example below is based on IOS 15.1(3)T.  Minor adjustments may be necessary for ealier IOS revisions.
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[[Cisco IOS|Cisco IOS Configuration]]
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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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<div style="width:100%;overflow:hidden;clear:both"></div>
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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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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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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
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  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
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  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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  no vad
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  voice-class sip bind media source-interface GigabitEthernet0/0      !Use your internet-facing interface here
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  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.
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!This is the first-priority dial peer for if your first-priority registrar server is available
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dial-peer voice 2 voip
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  destination-pattern T
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  no voice-class sip block 180
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  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:newyork.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
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  voice-class sip bind control source-interface GigabitEthernet0/0    !Use your internet-facing interface here
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  preference 1
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!This dial peer is the same as #2, but lower preference.
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!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.
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dial-peer voice 3 voip
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  destination-pattern T
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  no voice-class sip block 180
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  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
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  voice-class sip bind control source-interface GigabitEthernet0/0    !Use your internet-facing interface here
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  preference 2
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end
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copy run start
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<br><br><br><br><br>
<br><br><br><br><br>

Revision as of 17:21, 12 September 2014

Contents

FreePBX / PBX in a Flash (SIP)

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 (SIP) Configuration






FreePBX / PBX in a Flash (IAX2)

PBX in a Flash



If you've longed for the good ol' days of Asterisk@Home, welcome back to the new steroid-enhanced version. PBX in a Flash™ is the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users and VARs. You'll have a high-performance turnkey Asterisk PBX that's easy to upgrade with dozens of add on scripts to provide virtually any feature you can imagine. FreePBX / PBX in a Flash (IAX2) Configuration






Asterisk (SIP)

Asterisk



Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide. Asterisk (SIP) Configuration






Asterisk (IAX2)

Asterisk



Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide. Asterisk (IAX2) 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






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






3CX Phone System

3CX Phone System



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).

3CX Phone System Configuration






SIPfoundry

Sipfoundry-logo.png

SIPfoundry is the home of the sipXecs open source unified communications and collaboration project. If you are looking to replace your aging (IP) PBX with an all software modern communications solution that scales to mid-size and large enterprise, look no further. We are community dedicated to the development of SIP and XMPP based communications solutions. Founded in 2004 as an independent 501(3)c not-for-profit corporation based in Massachusetts, SIPfoundry is governed by an independent board of directors. SIPfoundry is open an invites all interested parties to cooperate and collaborate. While the sipXecs project is the largest active project at SIPfoundry, we are open to make available our infrastructure to other interested projects. A commercial enterprise edition of the sipXecs solution is available from eZuce with full support, software maintenance, training, indemnification, and other services.

To learn how to configure sipXecs to work with voip.ms, follow this 10 minute guide here:

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"/>







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






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






Personal tools
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Variants
Actions
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