ViciPhone
Posted: Thu Aug 01, 2019 6:49 pm
Hi,
I have installed Vicibox 8 and followed the instructions on the post in the Vicibox Install Forum (viewtopic.php?f=8&t=37686)
I have successfully setup SSL and https is working correctly.
I have also followed the instructions for the viciphone in the same post. I have tried both viciphone 1 and 2.
I am now trying to use viciphone but I am getting "req. failed" on the phone screen and I am getting the following error when I check the chrome console.
What is causing this?
Vicibox 8.1.2
VERSION: 2.14-717a
BUILD: 190724-1603
UPDATE: I have editted my WebRTC Phone Template and I am now able to register the phone.
However, I have no audio and I also keep getting "No one is in your session: 8600051"
Asterisk output is:
'server' is my hostname
I have installed Vicibox 8 and followed the instructions on the post in the Vicibox Install Forum (viewtopic.php?f=8&t=37686)
I have successfully setup SSL and https is working correctly.
I have also followed the instructions for the viciphone in the same post. I have tried both viciphone 1 and 2.
I am now trying to use viciphone but I am getting "req. failed" on the phone screen and I am getting the following error when I check the chrome console.
What is causing this?
Vicibox 8.1.2
VERSION: 2.14-717a
BUILD: 190724-1603
- Code: Select all
REGISTER sip:SERVERIP SIP/2.0
Via: SIP/2.0/TCP 192.0.2.64;branch=z9hG4bK4659604
Max-Forwards: 70
To: "9999" <sip:9999@SERVERIP>
From: "9999" <sip:9999@SERVERIP>;tag=4182v0bmpo
Call-ID: 66fmml5nvpjua9vc9am7s9
CSeq: 8485 REGISTER
Authorization: Digest algorithm=MD5, username="9999", realm="asterisk", nonce="0765e048", uri="sip:SERVERIP", response="b02632dbc6e5cb44d6e912b9324d65d0"
Contact: <sip:q1islpf8@192.0.2.64;transport=wss>;expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: VICIphone 2.0
Content-Length: 0
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 192.0.2.64;branch=z9hG4bK4659604;received=MYIP;rport=51287
From: "9999" <sip:9999@SERVERIP>;tag=4182v0bmpo
To: "9999" <sip:9999@SERVERIP>;tag=as42f29e1b
Call-ID: 66fmml5nvpjua9vc9am7s9
CSeq: 8485 REGISTER
Server: Asterisk PBX 13.21.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Date: Thu, 01 Aug 2019 23:36:57 GMT
Content-Length: 0
UPDATE: I have editted my WebRTC Phone Template and I am now able to register the phone.
However, I have no audio and I also keep getting "No one is in your session: 8600051"
Asterisk output is:
- Code: Select all
[Aug 2 01:10:58] == WebSocket connection from 'MYIP:57292' for protocol 'sip' accepted using version '13'
[Aug 2 01:10:58] == WebSocket connection from 'MYIP:56826' forcefully closed due to fatal write error
[Aug 2 01:10:58] -- Registered SIP '9998' at MYIP:57292
[Aug 2 01:11:04] ERROR[7694]: chan_sip.c:4270 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[Aug 2 01:11:05] == DTLS ECDH initialized (automatic), faster PFS enabled
[Aug 2 01:11:05] ERROR[15368]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("server", "(null)", ...): Name or service not known
[Aug 2 01:11:05] WARNING[15368]: acl.c:835 resolve_first: Unable to lookup 'server'
[Aug 2 01:11:05] == Using SIP RTP CoS mark 5
[Aug 2 01:11:05] -- Called 9998
'server' is my hostname