Asterisk PJSIP
From VoIP.ms Wiki
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===pjsip.conf=== | ===pjsip.conf=== | ||
| - | + | <nowiki> | |
| - | [ | + | [transport-udp] |
| - | + | type = transport | |
| + | protocol = udp | ||
| + | bind = 0.0.0.0 | ||
[voipms] | [voipms] | ||
| - | + | type = registration | |
| - | + | transport = transport-udp | |
| - | + | outbound_auth = voipms | |
| - | + | client_uri = sip:100000@atlanta.voip.ms:5060 ; (one of our multiple servers, you can choose the one closer to your location) | |
| - | type= | + | server_uri = sip:atlanta.voip.ms:5060 ; (one of our multiple servers, you can choose the one closer to your location) |
| - | username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) | + | |
| - | + | [voipms] | |
| - | + | type = auth | |
| - | + | auth_type = userpass | |
| - | + | username = 100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) | |
| - | + | password = johnspassword ; your password | |
| - | + | ||
| - | + | [voipms] | |
| - | + | type = aor | |
| + | contact = sip:[email protected] ; (one of our multiple servers, you can choose the one closer to your location) | ||
| + | |||
| + | [voipms] | ||
| + | type = endpoint | ||
| + | transport = transport-udp | ||
| + | context = mycontext | ||
| + | disallow = all | ||
| + | allow = ulaw | ||
| + | from_user = 100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) | ||
| + | auth = voipms | ||
| + | outbound_auth = voipms | ||
| + | aors = voipms | ||
| + | ; NAT parameters: | ||
| + | rtp_symmetric = yes | ||
| + | rewrite_contact = yes | ||
| + | send_rpid = yes | ||
| + | |||
| + | [voipms] | ||
| + | type = identify | ||
| + | endpoint = voipms | ||
| + | match = atlanta.voip.ms ; (one of our multiple servers, you can choose the one closer to your location) | ||
</nowiki> | </nowiki> | ||
Revision as of 13:20, 11 June 2015
Contents |
Asterisk (PJSIP)
pjsip.conf
[transport-udp] type = transport protocol = udp bind = 0.0.0.0 [voipms] type = registration transport = transport-udp outbound_auth = voipms client_uri = sip:[email protected]:5060 ; (one of our multiple servers, you can choose the one closer to your location) server_uri = sip:atlanta.voip.ms:5060 ; (one of our multiple servers, you can choose the one closer to your location) [voipms] type = auth auth_type = userpass username = 100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) password = johnspassword ; your password [voipms] type = aor contact = sip:[email protected] ; (one of our multiple servers, you can choose the one closer to your location) [voipms] type = endpoint transport = transport-udp context = mycontext disallow = all allow = ulaw from_user = 100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) auth = voipms outbound_auth = voipms aors = voipms ; NAT parameters: rtp_symmetric = yes rewrite_contact = yes send_rpid = yes [voipms] type = identify endpoint = voipms match = atlanta.voip.ms ; (one of our multiple servers, you can choose the one closer to your location)
- 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(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(PJSIP/${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. (PJSIP)
pjsip.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(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(PJSIP/${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
