Asterisk PJSIP

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[checked revision]

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Revision as of 20:36, 10 June 2015

Contents 1 Asterisk (SIP) 1.1 sip.conf 1.2 extensions.conf 2 Asterisk IP Auth. (SIP) 2.1 sip.conf 2.2 extensions.conf

Asterisk (SIP)

sip.conf

[general]                
register => 100000:[email protected]:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes

• 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 (one of our multiple servers, you can choose the one closer to your location)
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=yes

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