Webrtc PBXWebPhone problem

[andicrb76]

Dear All,

i have vicidial server, the specifications are:

VERSION: 2.14-781a
BUILD: 201214-1545


and the problem is everytime the agent login, after the webphone is registered and ready, when the customer data form show up,
the webphone terminate the callback, so the webphone cannot join the conference room,
when you should hear "youre the only person in the conference".

Anyone can help me ?
Here's the log:

== Manager 'sendcron' logged on from 127.0.0.1
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
Audio is at 18360
Adding codec opus to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 27.111.44.36:17114:
INVITE sip:s7413fp8@192.0.2.90;transport=wss SIP/2.0
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK18f989be
Max-Forwards: 70
From: "ACagcW16092306837001700170017001" <sip:0000000000@27.111.44.34>;tag=as4f2245ae
To: <sip:s7413fp8@192.0.2.90;transport=wss>
Contact: <sip:0000000000@27.111.44.34:5060;transport=ws>
Call-ID: 193a8a04655967d64f084daa2f105f54@27.111.44.34:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 29 Dec 2020 08:37:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "ACagcW16092306837001700170017001" <sip:0000000000@27.111.44.34>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 451

v=0
o=root 1407126552 1407126552 IN IP4 27.111.44.34
s=Asterisk PBX 13.21.0-vici
c=IN IP4 27.111.44.34
t=0 0
m=audio 18360 UDP/TLS/RTP/SAVPF 107 0 101
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 90:1E:A6:71:17:14:1E:3A:64:40:8C:34:72:C8:FF:15:F7:28:28:AE:C5:25:0B:89:19:F5:93:21:5B:32:80:7C
a=sendrecv

---
-- Called 7001

<--- SIP read from WS:27.111.44.36:17114 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK18f989be
To: <sip:s7413fp8@192.0.2.90;transport=wss>
From: "ACagcW16092306837001700170017001" <sip:0000000000@27.111.44.34>;tag=as4f2245ae
Call-ID: 193a8a04655967d64f084daa2f105f54@27.111.44.34:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:27.111.44.36:17114 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK18f989be
To: <sip:s7413fp8@192.0.2.90;transport=wss>;tag=k7ddsc7kq6
From: "ACagcW16092306837001700170017001" <sip:0000000000@27.111.44.34>;tag=as4f2245ae
Call-ID: 193a8a04655967d64f084daa2f105f54@27.111.44.34:5060
CSeq: 102 INVITE
Contact: <sip:s7413fp8@192.0.2.90;transport=wss>
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:s7413fp8@192.0.2.90;transport=wss>
-- SIP/7001-00000014 is ringing

<--- SIP read from WS:27.111.44.36:17114 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK18f989be
To: <sip:s7413fp8@192.0.2.90;transport=wss>;tag=k7ddsc7kq6
From: "ACagcW16092306837001700170017001" <sip:0000000000@27.111.44.34>;tag=as4f2245ae
Call-ID: 193a8a04655967d64f084daa2f105f54@27.111.44.34:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0



Thank you in advance

Andi

[carpenox]

have you have the webphone setup correctly at all yet? It doesnt sound like it. If you are not hearing the meetme sounds when you login there is a few things it can be, but most the time its that dahdi isnt running correctly or your missing some asterisk sockets. what happens when you run "dahdi_cfg -vvv" from linux?

[andicrb76]

i got response:

DAHDI Tools Version - 3.1.0
Notice: Configuration file is /etc/dahdi/system.conf
line 0: Unable to open configuration file '/etc/dahdi/system.conf'

1 error(s) detected


====================================================
VERSION: 2.14-781a | BUILD: 201214-1545 | Asterisk 13.21.0-vici | Single Server proxmox VM

[andicrb76]

after i reinstall dahdi module the response is:
[root@asterisk dahdi-linux-complete-2.9.1+2.9.1]# dahdi_cfg -vvv
DAHDI Tools Version - 2.9.1

DAHDI Version: 3.1.0
Echo Canceller(s):
Configuration
======================


Channel map:


0 channels to configure.

But, the callback still dropped but the log is different now, here's the log:

asterisk*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

<--- SIP read from WS:27.111.44.36:1821 --->
REGISTER sip:asterisk.d####.co.id SIP/2.0
Via: SIP/2.0/WSS 192.0.2.125;branch=z9hG4bK2187377
Max-Forwards: 70
To: "7001" <sip:7001@asterisk.d####.co.id>
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=bv53jtl7i8
Call-ID: ter4cmn4k04e5s67dqf530
CSeq: 3823 REGISTER
Authorization: Digest algorithm=MD5, username="7001", realm="asterisk.d####.co.id", nonce="361619a2", uri="sip:asterisk.d####.co.id", response="7c4744887111a5f086100c598172c9d1"
Contact: <sip:grjp7rt6@192.0.2.125;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:a938b94b-2812-4da8-9e2f-806a69337515>";expires=0
Supported: path, gruu, outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---

<--- Transmitting (no NAT) to 27.111.44.36:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.125;branch=z9hG4bK2187377;received=27.111.44.36
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=bv53jtl7i8
To: "7001" <sip:7001@asterisk.d####.co.id>;tag=as13c9dddb
Call-ID: ter4cmn4k04e5s67dqf530
CSeq: 3823 REGISTER
Server: Asterisk PBX 13.21.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.d####.co.id", nonce="3f8adf69"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ter4cmn4k04e5s67dqf530' in 32000 ms (Method: REGISTER)
== WebSocket connection from '27.111.44.36:14748' for protocol 'sip' accepted using version '13'

<--- SIP read from WS:27.111.44.36:14748 --->
REGISTER sip:asterisk.d####.co.id SIP/2.0
Via: SIP/2.0/WSS 192.0.2.191;branch=z9hG4bK8713121
Max-Forwards: 70
To: "7001" <sip:7001@asterisk.d####.co.id>
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=2u5vjcqpbd
Call-ID: ic8889om78ih3oecgj6jjr
CSeq: 1943 REGISTER
Contact: <sip:vonrkqh9@192.0.2.191;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:b9b721c6-df8f-4601-9395-7f7d0c21a2ef>";expires=30
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---

<--- Transmitting (no NAT) to 27.111.44.36:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.191;branch=z9hG4bK8713121;received=27.111.44.36
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=2u5vjcqpbd
To: "7001" <sip:7001@asterisk.d####.co.id>;tag=as65966b42
Call-ID: ic8889om78ih3oecgj6jjr
CSeq: 1943 REGISTER
Server: Asterisk PBX 13.21.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.d####.co.id", nonce="2a391f23"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ic8889om78ih3oecgj6jjr' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:27.111.44.36:14748 --->
REGISTER sip:asterisk.d####.co.id SIP/2.0
Via: SIP/2.0/WSS 192.0.2.191;branch=z9hG4bK8240386
Max-Forwards: 70
To: "7001" <sip:7001@asterisk.d####.co.id>
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=2u5vjcqpbd
Call-ID: ic8889om78ih3oecgj6jjr
CSeq: 1944 REGISTER
Authorization: Digest algorithm=MD5, username="7001", realm="asterisk.d####.co.id", nonce="2a391f23", uri="sip:asterisk.d####.co.id", response="526d86b6b81dc46dbc8f5d68f0b4f847"
Contact: <sip:vonrkqh9@192.0.2.191;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:b9b721c6-df8f-4601-9395-7f7d0c21a2ef>";expires=30
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
-- Registered SIP '7001' at 27.111.44.36:14748
Reliably Transmitting (no NAT) to 27.111.44.36:14748:
OPTIONS sip:vonrkqh9@192.0.2.191;transport=wss SIP/2.0
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK03070eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@27.111.44.34>;tag=as522499ef
To: <sip:vonrkqh9@192.0.2.191;transport=wss>
Contact: <sip:asterisk@27.111.44.34:5060;transport=ws>
Call-ID: 49f36e2b513bb40501f01ed47ed4d6b1@27.111.44.34:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Wed, 30 Dec 2020 02:35:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 27.111.44.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.191;branch=z9hG4bK8240386;received=27.111.44.36
From: "7001" <sip:7001@asterisk.d####.co.id>;tag=2u5vjcqpbd
To: "7001" <sip:7001@asterisk.d####.co.id>;tag=as65966b42
Call-ID: ic8889om78ih3oecgj6jjr
CSeq: 1944 REGISTER
Server: Asterisk PBX 13.21.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:vonrkqh9@192.0.2.191;transport=wss>;expires=60
Date: Wed, 30 Dec 2020 02:35:14 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7bec957512cc15404b5cb94b2f6ceb82@27.111.44.34:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 27.111.44.36:14748:
NOTIFY sip:vonrkqh9@192.0.2.191;transport=wss SIP/2.0
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK3768ae05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@27.111.44.34>;tag=as0b1fe834
To: <sip:vonrkqh9@192.0.2.191;transport=wss>
Contact: <sip:asterisk@27.111.44.34:5060;transport=ws>
Call-ID: 7bec957512cc15404b5cb94b2f6ceb82@27.111.44.34:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 13.21.0-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 107

Messages-Waiting: yes
Message-Account: sip:asterisk@27.111.44.34;transport=WS
Voice-Message: 46/0 (0/0)

---
Scheduling destruction of SIP dialog 'ic8889om78ih3oecgj6jjr' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:27.111.44.36:14748 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK03070eff
To: <sip:vonrkqh9@192.0.2.191;transport=wss>;tag=k7bd377vt1
From: "asterisk" <sip:asterisk@27.111.44.34>;tag=as522499ef
Call-ID: 49f36e2b513bb40501f01ed47ed4d6b1@27.111.44.34:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Accept: application/sdp,application/dtmf-relay
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- SIP read from WS:27.111.44.36:14748 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/WS 27.111.44.34:5060;branch=z9hG4bK3768ae05
To: <sip:vonrkqh9@192.0.2.191;transport=wss>;tag=eignphfmhl
From: "asterisk" <sip:asterisk@27.111.44.34>;tag=as0b1fe834
Call-ID: 7bec957512cc15404b5cb94b2f6ceb82@27.111.44.34:5060
CSeq: 102 NOTIFY
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0

[andicrb76]

update: sometimes i got 401 unauthorized and 480 Temporarily Unavailable error after ringing before call dropped

[carpenox]

how did you setup the template for the webphone? msg me on skype it will be easier to help you resolve this then we can post the resolution here for others incase they have the same problem