VoIP:Asterisk/Freepbx-No audio for established sip calls and call drops after 30 seconds 两条命令解决Asterisk通话无声音、30秒自动挂断的问题

2018年07月18日 2989Browse 2Like 0Comments


Asterisk 13.15.1: NAT enabled, SIP ports opened in firewall.


  1. calls can be established but no audio on both sides
  2. calls auto drop after 30 seconds


  1. firewall stopped RTP packets as RTP ports not opened: audio transmission needs UDP ports (10000-20000) opened;
  2. call will be cut if no RTP packets received in 30 seconds(this timeout can be customized in Chan SIP settings).


I Googled a lot and didn't get any clear guides or posts providing a solution. I solved it by adding a firewall rule when I found asterisk calls are over RTP ports(range is 10000-20000).

#firewall-cmd --zone=public --add-port=10000-20000/udp --permanent
#firewall-cmd --reload

Good luck!


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