Hello to all this community, this is my first post and I want to make this thread to all the people facing the same issue, I made a fresh install of vicidial with asterisk Version 13.34.0-vici and everything works fine with the actual setup, but my main problem is the WebRTC, we can make calls VIA sip to softphones, and we receive and send calls, i took down the firewall for WebRTC test, but anyways, i got no sound with my actual setup, I tried the same setup on AWS ec2 but no resolution to my case.
debug to webrtc phone
2020-10-06 20:29:51 =>
displayName: gs102
uri: gs102@10.128.0.8
authorizationUser: gs102
password: ****
wsServers: wss://FQDN:8089/ws
and seems almost promising but, i feel that i missing something and i hope you can help me with that, the phone registrations works but i got no sound and checking my asterisk console, i got this error
[Oct 6 22:30:03] WARNING[2269]: sip/config_parser.c:819 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
[Oct 6 22:30:03] WARNING[2269]: chan_sip.c:33172 reload_config: Cannot use 'tcp' transport with tcpenable=no. Removing from available transports.
[Oct 6 22:30:03] WARNING[2269]: chan_sip.c:33185 reload_config: No valid default transport. Selecting 'udp' as default.
[Oct 6 22:30:03] == Using SIP CoS mark 4
[Oct 6 22:30:03] WARNING[2269]: chan_sip.c:30880 set_insecure_flags: Unknown insecure mode 'very' on line 18
[Oct 6 22:30:03] WARNING[2269]: sip/config_parser.c:819 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
[Oct 6 22:30:07] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 6 22:30:07] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 6 22:30:16] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource)
[Oct 6 22:30:23] ERROR[2269]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Oct 6 22:30:23] ERROR[2269]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Oct 6 22:30:23] == WebSocket connection from '190.87.160.176:51050' forcefully closed due to fatal write error
And I tried to set up all step by step, I have knowledge about programming and values but I missing something, i will post the asterisk setup as well but resumed
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
enablestatic=yes
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/certbot/live/FQDN/cert.pem
tlsprivatekey=/etc/certbot/live/FQDN/privkey.pem
that’s the main setup, and the ssl certs works, i mean, when load the page it says that is secure, btw, I force all the connections to ssl, this is my sip.conf code, again, resumed
[general]
transport=tcp,udp,ws,wss
avpf=yes
udpbindaddr=0.0.0.0:5060
tcpbinaddr=0.0.0.0:5060
context=trunkinbound
allowguest=no
allowoverlap=no
realm=FQDN
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en
relaxdtmf=yes
trustrpid = no
sendrpid = yes
progressinband=no
videosupport=no
callevents=yes
dtmfmode = rfc2833
rtptimeout=60
notifyringing = yes ;
notifyhold = yes
externip = MYEXTERNALIP
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
nat=yes
canreinvite=no
jbenable = yes
jbforce = no
jbmaxsize = 100
jbresyncthreshold = 1000
jbimpl = fixed
jblog = no
qualify=yes
limitonpeer = yes
I used Amazon ec2 to have the same system, but now i’m using Google cloud, but is the same issue and i tried to fixed the two times, but no fix with the webrtc, i had setup the websocket server, but i think i’m missing some extra configuration that I will be glad if you can help me.
