How to solve the problem with voice transmission when calling with Asterisk?

Question

Asterisk

Asterisk on CentOS 6.8 on a virtual server in Alibaba Cloud.
The call through the SIP trunk is established, but RTP does not go. Obviously Asterisk is behind NAT.
In iptables RTP ports are allowed, selinux is disabled. Why there is no sound (and Zoiper writes 0kbps RTP on the smartphone) is not clear. What other steps can be taken for troubleshooting?
Maybe you need some special settings in the asterisk?

Answer 1

Problem solved by adding lines to sip.conf
externip = xx.xx.xx.xx
localnet = 192.168.1.0/255.255.255.0