Asterisk 呼叫时自动终止呼叫

Asterisk 呼叫时自动终止呼叫,asterisk,voip,sipml,Asterisk,Voip,Sipml,这是我的asterisk服务器控制台日志 [Feb 15 12:17:49]警告[3558][C-00000000]:res_rtp_asterisk.C:2141 dtlsetup:在“0x7fd64400caa0”上设置DTLS-SRTP时无法设置策略 [Feb 15 12:17:49]警告[3558][C-00000000]:res_rtp_星号。C:4465 ast_d:rtp读取错误:未指定。挂断 Channel SIP/7005-00000000 left 'simple_brid

这是我的asterisk服务器控制台日志

[Feb 15 12:17:49]警告[3558][C-00000000]:res_rtp_asterisk.C:2141 dtlsetup:在“0x7fd64400caa0”上设置DTLS-SRTP时无法设置策略

[Feb 15 12:17:49]警告[3558][C-00000000]:res_rtp_星号。C:4465 ast_d:rtp读取错误:未指定。挂断

 Channel SIP/7005-00000000 left 'simple_bridge' basic-bridge <222810-4890-bedf-84d549cea2b0>
读取错误:未指定


检查防火墙和NAT设置。

可能是SRTP编译问题
[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