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PBXs

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FreePBX / Trixbox / PBX in a Flash (SIP)

freepbxsiptrunk.gif

Pbx-left.jpg

canreinvite=nonat
; nat=yes ; uncomment if behind nat
context=from-trunk
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
qualify=yes


Register String

100000:johnspassword@atlanta.voip.ms:5060






FreePBX / Trixbox / PBX in a Flash (IAX2)

freepbxiax.gif

Pbx-left.jpg

type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=from-trunk
host=atlanta.voip.ms
disallow=all
allow=ulaw
insecure=port,invite
requirecalltoken=no
qualify=yes


Register String

100000:johnspassword@atlanta.voip.ms:4569






Asterisk (SIP)

sip.conf

[general]                
register => 100000:johnspassword@atlanta.voip.ms: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


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:johnspassword@atlanta.voip.ms

[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(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






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

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 Configuration

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