Automatic terminating a call when call made

Question

here is my Console log of asterisk server

[Feb 15 12:17:49] WARNING[3558][C-00000000]: res_rtp_asterisk.c:2141 dtlsetup: Could not set policies when setting up DTLS-SRTP on '0x7fd64400caa0

[Feb 15 12:17:49] WARNING[3558][C-00000000]: res_rtp_asterisk.c:4465 ast_d: RTP Read error: Unspecified. Hanging up.

 Channel SIP/7005-00000000 left 'simple_bridge' basic-bridge <222810-4890-bedf-84d549cea2b0>

== Spawn extension (default, 7008, 1) exited non-zero on 'SIP/7005-0000 -- Channel SIP/7008-00000001 left 'simple_bridge' basic-bridge <222810-4890-bedf-84d549cea2b0>

IS Any changes needed in sip.conf ? in sip.conf

[7005] ; This will be WebRTC client
type=peer ;
username=7005 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=Z-jj! ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws,wss,tcp ; Asterisk will allow this peer to register on UDP or WebSockets
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
disallow=all
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
dtlsverify=fingerprint
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
;nat=force_rport,comedia
force_avp=yes

Answer 1

Read error: Unspecified

Check your firewall and NAT settings.