WebRTC issue with vicibox fresh install
Posted: Wed Oct 31, 2018 2:24 pm
VERSION: 2.14-694a Installed with ViciBox_v8_1.x86_64-8.1.2
BUILD: 181005-1738
Having an issue with WebRTC:
viciphone debug log:
2018-10-31 15:21:53 =>
displayName: 2525
uri: 2525@66.58.85.146
authorizationUser: 2525
password: m5Y4FLQOUfROo2weeWWW
wsServers: wss://dialer.crft.com:8089/ws
2018-10-31 15:21:56 => Got Invite from <0000000000> "ACagcW15410136786666666666666666"
2018-10-31 15:21:56 => Auto-Answered Call
Call gets auto-answered on viciphone. But here's the CLI:
[Oct 31 15:12:26] == WebSocket connection from '12.333.333.65:49729' for protocol 'sip' accepted using version '13'
[Oct 31 15:12:26] -- Registered SIP '2525' at 12.XXX.XXX.65:49729
[Oct 31 15:12:26] NOTICE[3405]: chan_sip.c:24639 handle_response_peerpoke: Peer '2525' is now Reachable. (105ms / 2000ms)
[Oct 31 15:12:29] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 15:12:29] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 15:12:29] == Using SIP RTP CoS mark 5
[Oct 31 15:12:29] -- Called 2525
[Oct 31 15:12:29] -- SIP/2525-00000003 is ringing
asterisk is not detecting the answer.
Any ideas?
BUILD: 181005-1738
Having an issue with WebRTC:
viciphone debug log:
2018-10-31 15:21:53 =>
displayName: 2525
uri: 2525@66.58.85.146
authorizationUser: 2525
password: m5Y4FLQOUfROo2weeWWW
wsServers: wss://dialer.crft.com:8089/ws
2018-10-31 15:21:56 => Got Invite from <0000000000> "ACagcW15410136786666666666666666"
2018-10-31 15:21:56 => Auto-Answered Call
Call gets auto-answered on viciphone. But here's the CLI:
[Oct 31 15:12:26] == WebSocket connection from '12.333.333.65:49729' for protocol 'sip' accepted using version '13'
[Oct 31 15:12:26] -- Registered SIP '2525' at 12.XXX.XXX.65:49729
[Oct 31 15:12:26] NOTICE[3405]: chan_sip.c:24639 handle_response_peerpoke: Peer '2525' is now Reachable. (105ms / 2000ms)
[Oct 31 15:12:29] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 15:12:29] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 15:12:29] == Using SIP RTP CoS mark 5
[Oct 31 15:12:29] -- Called 2525
[Oct 31 15:12:29] -- SIP/2525-00000003 is ringing
asterisk is not detecting the answer.
Any ideas?