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Asterisk PJSIP

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(Created page with "==Asterisk (SIP)== ===sip.conf=== <nowiki> [general] register => 100000:[email protected]:5060 [voipms] canreinvite=no context=mycontext host=atla...")
(→pjsip.conf)
(8 intermediate revisions not shown)
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-
==Asterisk (SIP)==
+
==Asterisk (PJSIP)==
-
===sip.conf===
+
===pjsip.conf===
  <nowiki>
  <nowiki>
-
[general]              
+
[transport-udp]
-
register => 100000:johnspassword@atlanta.voip.ms:5060
+
type = transport
 +
protocol = udp
 +
bind = 0.0.0.0
[voipms]
[voipms]
-
canreinvite=no
+
type = registration
-
context=mycontext
+
transport = transport-udp
-
host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
+
outbound_auth = voipms
-
secret=johnspassword ;your password
+
client_uri = sip:100000@atlanta.voip.ms:5060    ; (one of our multiple servers, you can choose the one closer to your location)
-
type=peer
+
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)
+
-
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
+
-
</nowiki>
+
-
*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.
+
[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:100000@atlanta.voip.ms            ; (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
 +
; allow=g729                ; uncomment if you support g729
 +
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>
===extensions.conf===
===extensions.conf===
Line 34: Line 56:
[voipms-outbound]
[voipms-outbound]
-
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _1NXXNXXXXXX,n,Hangup()
-
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
+
exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,n,Hangup()
-
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _011.,n,Hangup()
-
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()
exten => _00.,n,Hangup()
Line 50: Line 72:
<br><br><br><br><br>
<br><br><br><br><br>
-
==Asterisk IP Auth. (SIP)==
+
==Asterisk IP Auth. (PJSIP)==
-
===sip.conf===
+
===pjsip.conf===
Note: You'll need to create a sub account to use IP Auth
Note: You'll need to create a sub account to use IP Auth
  <nowiki>
  <nowiki>
 +
[transport-udp]
 +
type = transport
 +
protocol = udp
 +
bind = 0.0.0.0
 +
[voipms]
[voipms]
-
canreinvite=nonat
+
type = aor
-
context=mycontext
+
contact = sip:100000@atlanta.voip.ms             ; (one of our multiple servers, you can choose the one closer to your location)
-
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>
+
 +
[voipms]
 +
type = endpoint
 +
transport = transport-udp
 +
context = mycontext
 +
disallow = all
 +
allow = ulaw
 +
; allow=g729                ; uncomment if you support g729
 +
from_user = 100000          ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
 +
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>
===extensions.conf===
===extensions.conf===
Line 77: Line 116:
[voipms-outbound]
[voipms-outbound]
-
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _1NXXNXXXXXX,n,Hangup()
-
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
+
exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,n,Hangup()
-
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _011.,n,Hangup()
-
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
+
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()
exten => _00.,n,Hangup()

Revision as of 13:40, 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
; allow=g729                 ; uncomment if you support g729
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)

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 

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0

[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
; allow=g729                 ; uncomment if you support g729
from_user = 100000           ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
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)

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
Retrieved from "https://wiki.voip.ms/article/Asterisk_PJSIP"
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