1. I am using asterisk-14.0.2 and it works well.I can use my voip phone to dial numbers

2. I followed the documentation at https://www.odoo.com/apps/modules/9.0/crm_voip/ to configure both asterisk server and odoo side,but I got some mistakes

3. It works well when I using voip to call someone,but once I answer the call,it will break quickly

4. Does anyone use asterisk to run odoo-voip?Please tell me how to solve it.

5. This is my configuration files:

sip.conf

[1050] ; This will be WebRTC client 
type=friend 
username=1050 ; The Auth user for SIP.js 
host=dynamic ; Allows any host to register 
secret=1050 ; 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=internal ; Tell Asterisk which context to use when this peer is dialing 
irectmedia=no ; Asterisk will relay media for this peer 
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets 
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 
dtlsenable=no ; Tell Asterisk to enable DTLS for this peer 
tlsverify=no ; Tell Asterisk to not verify your DTLS certs 
dtlscertfile=/etc/asterisk/keys/asterisk.crt ; 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 

extensions.conf

[internal] 
exten => 1050,1,Dial(SIP/1050) ; Dialing 1050 will call the SIP client registered to 1050 
exten => 1050,2,Hangup 
exten => _9.,1,Dial(dahdi/1/${EXTEN:1}) 
[incoming] 
exten => s,1,Dial(SIP/1050,20)
 exten => s,2,Hangup 

pjsip.conf

[transport-tls] 
type=transport 
protocol=tls 
bind=0.0.0.0:5061 
cert_file=/etc/asterisk/keys/asterisk.crt 
priv_key_file=/etc/asterisk/keys/asterisk.key 
method=tlsv1 

http.conf

[general] 
enabled=yes 
bindaddr=192.168.0.7 
bindport=8088