PBXs

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Contents 1 FreePBX / Trixbox / PBX in a Flash (SIP) 2 FreePBX / Trixbox / PBX in a Flash (IAX2) 3 Asterisk (SIP) 3.1 sip.conf 3.2 extensions.conf 4 Asterisk IP Auth. (SIP) 4.1 sip.conf 4.2 extensions.conf 5 Asterisk (IAX2) 5.1 iax.conf 5.2 extensions.conf 6 Cisco IOS 6.1 SIP Trunk (Username/Password Authentication)

FreePBX / Trixbox / PBX in a Flash (SIP)

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:[email protected]:5060

FreePBX / Trixbox / PBX in a Flash (IAX2)

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:[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

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(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
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