PBXs
From VoIP.ms Wiki
Contents |
FreePBX / PBX in a Flash (SIP)
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canreinvite=nonat |
Register String
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100000:[email protected]:5060 |
FreePBX / PBX in a Flash (IAX2)
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type=friend |
Register String
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100000:[email protected]:4569 |
Asterisk (SIP)
sip.conf
[general] register => 100000:[email protected]:5060 [voipms] canreinvite=no context=mycontext host=atlanta.voip.ms secret=johnspassword ;your password type=friend username=100000 ;your account disallow=all allow=ulaw ; allow=g729 ; Uncomment if you support G729 fromuser=100000 ;your account trustrpid=yes sendrpid=yes insecure=port,invite ; nat=yes ; Uncomment this if your box is behind a NAT
- 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.
extensions.conf
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound
[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()
; inbound context example for your DID numbers, do not add the number 1 in front
[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
Asterisk IP Auth. (SIP)
sip.conf
Note: You'll need to create a sub account to use IP Auth
[voipms] canreinvite=nonat context=mycontext host=atlanta.voip.ms type=friend disallow=all allow=ulaw ; allow=g729 ; uncomment if you support g729 insecure=port,invite ; nat=yes ; uncommment if behind a nat
extensions.conf
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound
[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()
; inbound context example for your DID numbers, do not add the number 1 in front
[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
Asterisk (IAX2)
iax.conf
register => 100000:[email protected] [voipms] type=friend username=100000 ;your account secret=johnspassword ;your password context=mycontext host=atlanta.voip.ms disallow=all allow=ulaw ; allow=g729 ; uncomment if you support it insecure=port,invite requirecalltoken=no
extensions.conf
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound
[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(IAX2/voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(IAX2/voipms/1${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(IAX2/voipms/${EXTEN})
exten => _011.,n,Hangup()
exten => _00.,1,Dial(IAX2/voipms/${EXTEN})
exten => _00.,n,Hangup()
; inbound context example for your DID numbers, do not add the number 1 in front
[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
Cisco IOS
SIP Trunk (Username/Password Authentication)
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 #.
configure terminal voice service voip gcid clid substitute name allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip e911 transport switch udp tcp asserted-id ppi localhost dns:dns.name.of.your.device midcall-signaling passthru no call service stop sip-ua credentials username your_account password 0 your_password realm voip.ms authentication username your_account password 0 your_password registrar 1 dns:newyork.voip.ms !Pick your preferred first server registrar 2 dns:montreal.voip.ms !Pick the next best here !You can configure up to 6 registrar servers for fault-tolerance !This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999 dial-peer voice 1 voip incoming called-number 5555551... voice-class sip asserted-id ppi no voice-class sip block 180 no voice-class sip block 181 no voice-class sip block 183 voice-class sip pass-thru headers unsupp voice-class sip pass-thru content unsupp voice-class sip pass-thru content sdp dtmf-relay rtp-nte !This dial peer is for outgoing calls and matches anything. !Finish dialing with a # to immediately route the call. dial-peer voice 2 voip destination-pattern T voice-class sip asserted-id ppi no voice-class sip block 180 no voice-class sip block 181 no voice-class sip block 183 voice-class sip pass-thru headers unsupp voice-class sip pass-thru content unsupp voice-class sip pass-thru content sdp dtmf-relay rtp-nte session protocol sipv2 session transport udp session target sip-server end copy run start
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.
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).
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 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:
http://blog.myitdepartment.net/?p=191
PBXES.ORG
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.
