Asterisk SIP
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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=== | ===extensions.conf=== | ||
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<br><br><br><br><br> | <br><br><br><br><br> | ||
| - | + | ==Asterisk IP Auth. (SIP)== | |
| + | |||
| + | ===sip.conf=== | ||
| + | |||
| + | Note: You'll need to create a sub account to use IP Auth | ||
| + | |||
| + | <nowiki> | ||
| + | [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 | ||
| + | </nowiki> | ||
| + | |||
===extensions.conf=== | ===extensions.conf=== | ||
Revision as of 17:06, 12 September 2014
Contents |
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
