Asterisk SIP

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[quality revision]

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Revision as of 18:57, 18 September 2014 (view source) Jeff (Talk | contribs) ← Older edit		Latest revision as of 16:13, 19 November 2019 (view source) Edward (Talk | contribs) (→Asterisk TLS/SRTP (SIP))
(9 intermediate revisions not shown)
Line 10:		Line 10:
	canreinvite=no		canreinvite=no
	context=mycontext		context=mycontext
-	host=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)
	secret=johnspassword ;your password		secret=johnspassword ;your password
	type=peer		type=peer
-	username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)	+	username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
	disallow=all		disallow=all
	allow=ulaw		allow=ulaw
	; allow=g729 ; Uncomment if you support G729		; 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)	+	fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
	trustrpid=yes		trustrpid=yes
	sendrpid=yes		sendrpid=yes
Line 48:		Line 48:
	exten => 7863643011,1,Answer() ;your DID		exten => 7863643011,1,Answer() ;your DID
	</nowiki>		</nowiki>
-	<br><br><br><br><br>	+	<br>
		+
		+	==Asterisk TLS/SRTP (SIP)==
		+
		+
		+	1. In order to use these devices with encryption, besides having to [[Call Encryption - TLS/SRTP | enable the SIP account in your VoIP.ms customer portal]], there are some settings you will have to modify in your device's configuration.
		+
		+	2. Once your account/sub-account has Encrypted traffic enabled, the system has to be configured use/send the traffic through TLS and SRTP.
		+
		+
		+	For the registration over TLS, you need to define the protocol the PBX will use in the general config.
		+
		+	[general]
		+	register => tls://100000:johnspassword@atlanta1.voip.ms:5061
		+
		+
		+	For the outbound part, add the following lines to the peer details.
		+
		+	encryption=yes
		+	transport=tls
		+
		+	See example below:
		+
		+	<nowiki>
		+	[voipms]
		+	encryption=yes
		+	transport=tls
		+	canreinvite=no
		+	context=mycontext
		+	host=atlanta1.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
		+	</nowiki>
		+
		+	'''Note:''' When using TLS is very important to specify the number of the server, in case the name you have chosen doesn't use the number 1 you need to add it.
	==Asterisk IP Auth. (SIP)==		==Asterisk IP Auth. (SIP)==
Line 60:		Line 103:
	canreinvite=nonat		canreinvite=nonat
	context=mycontext		context=mycontext
-	host=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		type=peer
	disallow=all		disallow=all
Line 101:		Line 144:
	[general] Section		[general] Section
		+
	mwi => 123456:mypassword@losangeles.voip.ms/65000		mwi => 123456:mypassword@losangeles.voip.ms/65000
	Add this Line please.		Add this Line please.
-	[voipms]Section	+	[voipms] Section
		+
	unsolicited_mailbox=65000		unsolicited_mailbox=65000
		+
		+
		+	== Local extension calling ==
		+
		+	(User submitted) In order for this to work, you must fill out "User Context and User Details"
		+
		+
		+
		+	User Context Name: 100000 (your account)
		+
		+	[100000] your account
		+	type=user
		+	auth=md5
		+	notransfer=yes
		+	disallow=all
		+	allow=gsm&ulaw
		+	trunk=yes
		+	secret=****
		+	context=from-trunk
	<br><br><br><br><br>		<br><br><br><br><br>

---

Latest revision as of 16:13, 19 November 2019

Contents 1 Asterisk (SIP) 1.1 sip.conf 1.2 extensions.conf 2 Asterisk TLS/SRTP (SIP) 3 Asterisk IP Auth. (SIP) 3.1 sip.conf 3.2 extensions.conf 4 Known Issues 4.1 Getting Message Waiting Indicator to Work with VoIP.ms Voicemail 5 Local extension calling

Asterisk (SIP)

sip.conf

[general]                
register => 100000:johnspassword@atlanta.voip.ms: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 TLS/SRTP (SIP)

1. In order to use these devices with encryption, besides having to enable the SIP account in your VoIP.ms customer portal, there are some settings you will have to modify in your device's configuration.

2. Once your account/sub-account has Encrypted traffic enabled, the system has to be configured use/send the traffic through TLS and SRTP.

For the registration over TLS, you need to define the protocol the PBX will use in the general config.

[general]                
register => tls://100000:johnspassword@atlanta1.voip.ms:5061

For the outbound part, add the following lines to the peer details.

encryption=yes
transport=tls

See example below:

[voipms]
encryption=yes
transport=tls
canreinvite=no
context=mycontext
host=atlanta1.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: When using TLS is very important to specify the number of the server, in case the name you have chosen doesn't use the number 1 you need to add it.

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

Known Issues

Getting Message Waiting Indicator to Work with VoIP.ms Voicemail

VoIP.ms sends MWI notifications unsolicited. Please make the following changes in your configurations.

Remove this Line please.

[general] Section

mwi => 123456:mypassword@losangeles.voip.ms/65000

Add this Line please.

[voipms] Section

unsolicited_mailbox=65000

Local extension calling

(User submitted) In order for this to work, you must fill out "User Context and User Details"

User Context Name: 100000 (your account)

[100000] your account
type=user
auth=md5
notransfer=yes
disallow=all
allow=gsm&ulaw
trunk=yes
secret=****
context=from-trunk