https://starkautosales.com/auction/audiotest.php (not-in-use)
https://provision.wuyifan.com/audiotest.php
https://sipjs.com/guides/make-call/
Asterisk Log:
== WebSocket connection from '10.11.1.10:59222' for protocol 'sip' accepted using version '13'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [660@from-internal:1] GotoIf("SIP/264-00000006", "1?ext-local,660,1") in new stack
Asterisk config file:
[root@webrtc asterisk]# more http_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[general]
enabled=yes
enablestatic=no
bindaddr=::
bindport=8088
prefix=
tlsenable=yes
tlsbindport=8089
tlsbindaddr=::
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=
| pbx6 custom destination | pbx6 dialplan |
| adminconf,s,1 | [adminconf] exten => s,1,Authenticate(2444,) exten => s,n,MeetMe(660,a) |
| pbx6 incoming route | pbx6 conference |
| go custom destination | 660 (all no, except, yes for “mute on join” |