Configuring Asterisk to Use Asterphone with PJSIP

Global settings

Settings - Asterisk SIP settings

SIP settings (chan_pjsip)
Misc PJSip Settings	taskprocessor overload trigger = pjsip_only
TLS/SSL/SRTP Settings	certificate manager = example.simple.test ssl method = tlsv1_2
wss	wss - 0.0.0.0 - All = No wss - 10.0.0.133 - ens18 = Yes

Settings - Advanced Settings

Asterisk Builtin mini-HTTP server
Force WebSocket Mode = pjsip

Add extension

Connectivity - Add Extension - Add New [chan_pjsip] Extension

General
Add Extension	extension = 100 (internal phone number) display name = 100 (must match the extension - the number that will be displayed when the call is made) secret - password, filled in automatically, required field, you can enter your own

Advanced

Add Extension

enable avpf = true if you leave it set to “no,” the connection is established successfully, and the call reaches the recipient, but is immediately disconnected enable ICESupport = true if you leave it set to “no,” the connection is established successfully, and the call reaches the recipient, but is immediately disconnected enable rtcpMux = true if you leave it set to “no,” the connection is established successfully, and the call reaches the recipient, but is immediately disconnected media encryption = dtls-srtp (not recommended) сonfiguring media stream encryption enable dtls = yes use encryption use certificate <example>.com (select current) select a certificate

Tab Advanced → Section Edit Extension
Transport	All - WSS Primary
Enable AVPF	Yes
Force AVP	Yes
Enable ICE Support	Yes
Enable rtcp Mux	Yes
Enable Encryption	Yes (SRTP only)
Section DTLS
Enable DTLS	Yes (there should be)
Use Certificate	<example>.com