Configuring Asterisk to Use Asterphone with PJSIP

Global settings

Settings - Asterisk SIP settings

SIP settings (chan_pjsip)
Misc PJSip Settingstaskprocessor overload trigger = pjsip_only
TLS/SSL/SRTP Settingscertificate manager = example.simple.test
ssl method = tlsv1_2
wsswss - 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 Extensionextension = 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 Extensionenable 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
TransportAll - WSS Primary
Enable AVPFYes
Force AVPYes
Enable ICE SupportYes
Enable rtcp MuxYes
Enable EncryptionYes (SRTP only)
Section DTLS
Enable DTLSYes (there should be)
Use Certificate&lt;example>.com

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