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: firstname.lastname@example.org 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: 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: chan_sip.c:33172 reload_config: Cannot use 'tcp' transport with tcpenable=no. Removing from available transports. [Oct 6 22:30:03] WARNING: 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: chan_sip.c:30880 set_insecure_flags: Unknown insecure mode 'very' on line 18 [Oct 6 22:30:03] WARNING: 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: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor [Oct 6 22:30:23] ERROR: 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 '22.214.171.124:51050' 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.