Asterisk PJSIP

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[draft revision]

[draft revision]

Revision as of 13:22, 11 June 2015 (view source) Michel (Talk | contribs) (→pjsip.conf) ← Older edit		Revision as of 13:22, 11 June 2015 (view source) Michel (Talk | contribs) (→pjsip.conf) Newer edit →
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	[voipms-inbound]		[voipms-inbound]
	exten => 7863643011,1,Answer() ;your DID		exten => 7863643011,1,Answer() ;your DID
		+
		+
		+	[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)
		+
		+
		+
	</nowiki>		</nowiki>
	<br><br><br><br><br>		<br><br><br><br><br>

---

Revision as of 13:22, 11 June 2015

Contents 1 Asterisk (PJSIP) 1.1 pjsip.conf 1.2 extensions.conf 2 Asterisk IP Auth. (PJSIP) 2.1 pjsip.conf 2.2 extensions.conf

Asterisk (PJSIP)

pjsip.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


[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)

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