Well if both of the previous steps don’t work for you. Then check if you are in a local network? If Yes, then you will need to add additional parameters in /etc/asterisk/sip.conf. Under the general context add your Public IP, NAT and local IP. Example [general] externip= (Your public ip) localnet= (your local network address) nat = yes
Howto setup Asterisk behind NAT | SYSadmin.lk The configuration option nat must be set to yes, and you may want to set qualify to yes as well although not necessary. With these steps, when properly configured, your external device should be able to communicate with your Asterisk PBX server unless you have issues on the remote end where the device is located because of badly behaved Firewalls. Asterisk Guru Asterisk Guru Website. 1.1 Description of the problem:. Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio).
NAT in VoIP - Cisco
If your Asterisk PBX is behind a NAT firewall, i.e. the PBX has an IP such as 192.168.0.2 then you will need to perform additional configuration to allow Asterisk to route the SIP and RTP correctly. The NAT configuration can be found in the file /etc/asterisk/sip.conf, the relevant section that needs to be edited is reproduced below:
Nat=yes . The NAT option determines the type of setting for users trying to connect to an asterisk server. With VoIP.ms it is recommended to have the NAT option set on Yes, which is the option that will work best. Portal settings influencing NAT with Asterisk: yes = Always ignore info and assume NAT; no = Use NAT mode only according to RFC3581
The configuration option nat must be set to yes, and you may want to set qualify to yes as well although not necessary. With these steps, when properly configured, your external device should be able to communicate with your Asterisk PBX server unless you have issues on the remote end where the device is located because of badly behaved Firewalls. Asterisk Guru Asterisk Guru Website. 1.1 Description of the problem:. Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). No audio on Asterisk SIP call - Stack Overflow