sipML5 and webRTC, Forbidden (403) error

[zamaniramin549]

Hi everyone, I have managed to install asterisk on VM and when I use sip software phone can make and receive calls, I would like to implement for web (webRTC) but after doing all the settings in asterisk documentation I get Forbidden (403) error, I was wondering if someone could help me,

Here is my console log:

WebSocket supported = yes
tsk_utils.js?svn=241:116 Navigator friendly name = chrome
tsk_utils.js?svn=241:116 OS friendly name = windows
tsk_utils.js?svn=241:116 Have WebRTC = yes
tsk_utils.js?svn=241:116 Have GUM = yes
tsk_utils.js?svn=241:116 Engine initialized
tsk_utils.js?svn=241:116 s_websocket_server_url=wss://pbx.lizard.global:8089/ws
tsk_utils.js?svn=241:116 s_sip_outboundproxy_url=(null)
tsk_utils.js?svn=241:116 b_rtcweb_breaker_enabled=no
tsk_utils.js?svn=241:116 b_click2call_enabled=no
tsk_utils.js?svn=241:116 b_early_ims=no
tsk_utils.js?svn=241:116 b_enable_media_stream_cache=yes
tsk_utils.js?svn=241:116 o_bandwidth={}
tsk_utils.js?svn=241:116 o_video_size={}
tsk_utils.js?svn=241:116 SIP stack start: proxy=‘ns313841.ovh.net:11060’, realm=‘sip:asterisk.org’, impi=‘1004’, impu=’“1004"sip:1004@pbx.lizard.global’
tsk_utils.js?svn=241:116 Connecting to ‘wss://pbx.lizard.global:8089/ws’
tsk_utils.js?svn=241:116 ==stack event = starting
tsk_utils.js?svn=241:116 __tsip_transport_ws_onopen
tsk_utils.js?svn=241:116 ==stack event = started
tsk_utils.js?svn=241:116 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
tsk_utils.js?svn=241:116 SEND: REGISTER sip:asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK6BrqPJcHebzuCPQwvZ6OF9rHhS6FZ7K1;rport
From: “1004"sip:1004@pbx.lizard.global;tag=nlQRy4gz1MaMqidg2wTh
To: “1004"sip:1004@pbx.lizard.global
Contact: “1004"sips:1004@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr”
Call-ID: be0314ca-1e4e-c873-41a8-6a8a1c5745b0
CSeq: 18178 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1004”,realm=“asterisk.org”,nonce=””,uri=“sip:asterisk.org”,response=””
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path

tsk_utils.js?svn=241:116 ==session event = connecting
tsk_utils.js?svn=241:116 ==session event = sent_request
tsk_utils.js?svn=241:116 __tsip_transport_ws_onmessage
tsk_utils.js?svn=241:116 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=59169;received=175.142.62.115;branch=z9hG4bK6BrqPJcHebzuCPQwvZ6OF9rHhS6FZ7K1
From: "1004"sip:1004@pbx.lizard.global;tag=nlQRy4gz1MaMqidg2wTh
To: "1004"sip:1004@pbx.lizard.global;tag=as59a8e502
Call-ID: be0314ca-1e4e-c873-41a8-6a8a1c5745b0
CSeq: 18178 REGISTER
Content-Length: 0
Server: FPBX-15.0.17.67(16.10.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm=“asterisk”,nonce=“2761070f”,stale=FALSE,algorithm=MD5

tsk_utils.js?svn=241:116 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
tsk_utils.js?svn=241:116 SEND: REGISTER sip:asterisk.org SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKwfJr7zWOBIgdDiS5I4PscNAbhRLKKxdP;rport
From: "1004"sip:1004@pbx.lizard.global;tag=nlQRy4gz1MaMqidg2wTh
To: "1004"sip:1004@pbx.lizard.global
Contact: "1004"sips:1004@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr”
Call-ID: be0314ca-1e4e-c873-41a8-6a8a1c5745b0
CSeq: 18179 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1004”,realm=“asterisk”,nonce=“2761070f”,uri=“sip:asterisk.org”,response=“a9aca512f65407cef40b5cd21759df50”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path

tsk_utils.js?svn=241:116 ==session event = sent_request
tsk_utils.js?svn=241:116 __tsip_transport_ws_onmessage
tsk_utils.js?svn=241:116 recv=SIP/2.0 403 Forbidden
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=59169;received=175.142.62.115;branch=z9hG4bKwfJr7zWOBIgdDiS5I4PscNAbhRLKKxdP
From: "1004"sip:1004@pbx.lizard.global;tag=nlQRy4gz1MaMqidg2wTh
To: "1004"sip:1004@pbx.lizard.global;tag=as59a8e502
Call-ID: be0314ca-1e4e-c873-41a8-6a8a1c5745b0
CSeq: 18179 REGISTER
Content-Length: 0
Server: FPBX-15.0.17.67(16.10.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

tsk_utils.js?svn=241:116 State machine: tsip_dialog_register_Any_2_Terminated_X_Error
tsk_utils.js?svn=241:116 === REGISTER Dialog terminated ===
tsk_utils.js?svn=241:116 ==session event = terminated

[david551]

You will need to provide the logging from the Asterisk side to be able to determine whether this is anything other than the, obvious, wrong password.

Also note that you appear to be using FreePBX. Any advice you receive here on the Asterisk configuration will be in terms of directly manipulating the .conf files, not in terms of the GUI.

You haven’t said which version of Asterisk, or which channel driver you are using.

[zamaniramin549]

and the asterisk version:
Asterisk 16.10.0 built by root @ freepbx777 on a x86_64 running Linux on 2020-05-15 11:03:48 UTC

[david551]

The log is not useful, as it does not show anything about the failed attempt. Also, we expect logs as plain text, and a useful log is almost always many screenfuls.

See Collecting Debug Information - Asterisk Project - Asterisk Project Wiki

And given you are running FreePBX, see Providing Great Debug - Support Services - Documentation

I’d also mention that Joshua Colp would say that you need to be very familiar with analysing logs for half a dozen technologies to be able to successfully use WebRTC. It is not plug and play.

[zamaniramin549]

I know it’s not plug and play and that’s why I’m here to troubleshoot.