Sip not registered - Wrong Password!

hi gays.
i can connect to my freePbx server with jssip module from nodeJs.
but registeration got faild with wrong password in Asterisk Logs.
i can connect and register with none WebRtc and WebSocket clients with same password for my PjSip Extension.

the Asterisk Log:

ERROR[24788]: res_http_websocket.c:506 ws_safe_read: Error readng from webScoket: Connection rest by peer.
NOTICE[33279]: chan_sip.c28486 handle_request_register: Registeration from ‘sip:[email protected]’ faild for - wrong password

jssip conf:

var socket = new JsSIP.WebSocketInterface(‘ws://’);

var configuration = {
sockets : [ socket ],
authorization_user: ‘170’,
uri : ‘sip:[email protected]’,
ws_servers : ‘ws://’,
password : ‘856589’,
realm : ‘’,
display_name : ‘170’,
contact_uri : ‘sip:[email protected]

jssip log:

Debugger listening on ws://
2017-11-09T19:07:51.269Z JsSIP:UA new() [configuration:{ sockets: [ NodeWebSocket { _url: ‘ws://’, _options: {}, _sipUri: ‘sip:;transport=ws’, _viaTransport: ‘WS’, _ws: null } ], authorization_user: ‘171’, uri: ‘sip:[email protected]’, wsServers: ‘ws://’, password: ‘secure’, realm: ‘’, display_name: ‘171’, contact_uri: ‘sip:[email protected]’ }]
2017-11-09T19:07:51.331Z JsSIP:Transport new()
2017-11-09T19:07:51.334Z JsSIP:UA configuration parameters after validation:
2017-11-09T19:07:51.334Z JsSIP:UA - authorization_user: “170"
2017-11-09T19:07:51.335Z JsSIP:UA - password: NOT SHOWN
2017-11-09T19:07:51.335Z JsSIP:UA - realm: “"
2017-11-09T19:07:51.335Z JsSIP:UA - ha1: NOT SHOWN
2017-11-09T19:07:51.335Z JsSIP:UA - display_name: “170"
2017-11-09T19:07:51.336Z JsSIP:UA - uri: sip:[email protected]
2017-11-09T19:07:51.336Z JsSIP:UA - contact_uri: {”_parameters”:{},”_headers":{},"_scheme":“sip”,"_user":“171”,"_host":“”}
2017-11-09T19:07:51.336Z JsSIP:UA - instance_id: "4506242c-c027-4b22-8be5-f15af615054d"
2017-11-09T19:07:51.336Z JsSIP:UA - use_preloaded_route: false
2017-11-09T19:07:51.337Z JsSIP:UA - session_timers: true
2017-11-09T19:07:51.337Z JsSIP:UA - no_answer_timeout: 60000
2017-11-09T19:07:51.337Z JsSIP:UA - register: true
2017-11-09T19:07:51.337Z JsSIP:UA - register_expires: 600
2017-11-09T19:07:51.337Z JsSIP:UA - registrar_server: sip:
2017-11-09T19:07:51.338Z JsSIP:UA - connection_recovery_max_interval: null
2017-11-09T19:07:51.338Z JsSIP:UA - connection_recovery_min_interval: null
2017-11-09T19:07:51.338Z JsSIP:UA - via_host: ""
2017-11-09T19:07:51.339Z JsSIP:UA start()
2017-11-09T19:07:51.340Z JsSIP:Transport connect()
Object {response: IncomingResponse, cause: “Rejected”}
response:IncomingResponse {data: “SIP/2.0 403 Forbidden\r\nVia: SIP/2.0/WS
”, headers: Object, method: “REGISTER”,

I am also trying to implement SIP client for my web application. Did your find solution for this problem ?

When I try to register with CTXSIP Web client

On FreePBX server PC, Asterisk showing the following error ?

handle_tcptls_connection: Problem setting up ssl connection: error: 00000001: lib(0);func(0):reason(1), Internal SSL error

I am using self-signed SSL certificate (default certificate available at FreePBX)
How to solve this error. Is it need any certificate needed or any settings needs to be done ?