Asterisk SIP

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[quality revision]

[draft revision]

Revision as of 17:49, 17 July 2015 (view source) Rod (Talk | contribs) ← Older edit		Revision as of 15:34, 19 November 2019 (view source) Edward (Talk | contribs) Newer edit →
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	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. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at '''Main Menu>> Account settings>> Advanced tab''' and for a sub-account at '''Sub accounts>> Manage sub-accounts''' and by clicking on the orange icon with a pen and click at "Advanced Options  Click here to display"
		+
		+	[[File:Mainacc encryp.png|thumb|none|300px|Click to enlarge]]
		+
		+	[[File:Subacc encryp.png|thumb|none|300px|Click to enlarge]]
		+
		+	2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.
		+
		+
		+	For the registration over TLS, you need to define the protocol the PBX will use in the general config.
		+
		+	<nowiki>
		+	[general]
		+	register => tls://100000:johnspassword@atlanta.voip.ms:5061
		+	</nowiki>
		+
		+	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=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
		+	</nowiki>
		+
	==Asterisk IP Auth. (SIP)==		==Asterisk IP Auth. (SIP)==

---

Revision as of 15:34, 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. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at Main Menu>> Account settings>> Advanced tab and for a sub-account at Sub accounts>> Manage sub-accounts and by clicking on the orange icon with a pen and click at "Advanced Options Click here to display"

2. Now that your account/sub-account has this setting enabled, your device only needs to send 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@atlanta.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=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

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