airsay Posted May 3, 2023 Report Share Posted May 3, 2023 @mc_ I have Kazoo on a Centos 7 box behind my ISP router. So definitely NATted. Softphones outside my network are able to register to the Kazoo box as I have open the following ports and forwarded them to the Kazoo box 5060 (tcp and udp) 7000 (tcp and udp) 10,000 - 60,000 (udp) I have red this thread from 2018 and updated the sipinterface_1.xml (located at /etc/kazoo/freeswitch/sip_profiles/) as advised. But I still don't have audio for calls from softphones outside my LAN. Any advice is welcome Link to comment Share on other sites More sharing options...
airsay Posted May 4, 2023 Author Report Share Posted May 4, 2023 Finally fixed this using advise from here. I've also included a screenshot in the event that someone (or myself) comes in the future and the referenced link is down. Long story short, you'd need to edit the following lines listen=UDP_SIP listen=TCP_SIP at the very bottom of /etc/kazoo/kamailio/local.cfg to read listen=UDP_SIP advertise YOUR_EXTERNAL_IP:5060 listen=TCP_SIP advertise YOUR_EXTERNAL_IP:5060 then restart kamailio with systemctl restart kazoo-kamailio Link to comment Share on other sites More sharing options...
2600Hz Employees mc_ Posted May 4, 2023 2600Hz Employees Report Share Posted May 4, 2023 Yes, running KAZOO in NAT'd environments requires this. I wonder how many clusters run like this? Link to comment Share on other sites More sharing options...
airsay Posted May 4, 2023 Author Report Share Posted May 4, 2023 This is a homelab test environment which is why I have the server behind a NAT. It resolved audio issues for calls from extensions outside my LAN to extensions inside the LAN. Calls from extensions inside the LAN to extensions outside the LAN still lack audio. Any pointers how to resolve this? Link to comment Share on other sites More sharing options...
2600Hz Employees mc_ Posted May 4, 2023 2600Hz Employees Report Share Posted May 4, 2023 Check the SDP on the legs. FreeSWITCH should use your ext-sip-ip and ext-rtp-ip if bridging direct to the device over the WAN in the SDP. Link to comment Share on other sites More sharing options...
airsay Posted May 5, 2023 Author Report Share Posted May 5, 2023 (edited) So with the above configuration, I have run into a problem and need some advise. Calls from extensions outside the LAN to an extension inside the LAN have one way audio. The extension outside the LAN can hear the callee within the LAN but the callee inside the LAN does not hear the caller from outside the LAN. Calls from extension inside the LAN to an extension outside the LAN have no audio both ways. Calls from extension outside the LAN to another extension outside the LAN also have no audio both ways In both situations, calls doesn't hang up after 32 seconds. On /etc/kazoo/freeswitch/sip_profiles/sipinterface_1.xml I have the following settings. <!-- SIP --> <param name="sip-ip" value="$${local_ip_v4"> <param name="ext-sip-ip" value="MY_EXT_IP"/> <!-- Media --> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="MY_EXT_IP"/> Completely lost on how to go on Edited May 5, 2023 by airsay (see edit history) Link to comment Share on other sites More sharing options...
2600Hz Employees mc_ Posted May 5, 2023 2600Hz Employees Report Share Posted May 5, 2023 Check the INVITE and 200 packets' SDP payloads, verify the correct IP is present for FreeSWITCH's packet at least. Link to comment Share on other sites More sharing options...
fmateo05 Posted May 6, 2023 Report Share Posted May 6, 2023 22 hours ago, airsay said: So with the above configuration, I have run into a problem and need some advise. Calls from extensions outside the LAN to an extension inside the LAN have one way audio. The extension outside the LAN can hear the callee within the LAN but the callee inside the LAN does not hear the caller from outside the LAN. Calls from extension inside the LAN to an extension outside the LAN have no audio both ways. Calls from extension outside the LAN to another extension outside the LAN also have no audio both ways In both situations, calls doesn't hang up after 32 seconds. On /etc/kazoo/freeswitch/sip_profiles/sipinterface_1.xml I have the following settings. <!-- SIP --> <param name="sip-ip" value="$${local_ip_v4"> <param name="ext-sip-ip" value="MY_EXT_IP"/> <!-- Media --> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="MY_EXT_IP"/> Completely lost on how to go on Can you try by uncommenting <param name="aggressive-nat-detection" value="true"/> ? Link to comment Share on other sites More sharing options...
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