The setup: 97 year old refinery with some bad phone cables (imagine that!) Client is moving to VoIP in the next 18 months. Looking to bypass bad lines with voip connections feeding through Asterisk to GXW4108. Current config on GXw4108 & FreePBX located here (pdf with screenshots) https://tinyurl.com/y8otmxnj
Configuation: Lucent Definity phone system from the very early 80s. About half a ton of soon to be junk. The Definity POTS extensions will be routed through the GXW4108 as needed to bypass bad phone lines. The network is ‘flat’, no routers/gateways. Consecutive IP addresses between all devices.
Inbound calls from the Definity through the GXW4108 work perfect. Disconnect works as expected. Outbound calls from the HT701 through Freepbx route to the GXW4108 and DO ring the proper extension on the Definity. However, there is NO ringback tone to the HT701 unless FreePBX generates it with Ttr or Tr options. Once the extension on the Definity is picked up, there is NO audio.
I’ve tried SIP and PJSIP connections, SIP has NO audio in either direction. PJSIP halfway works as above. I’ve tried PJSIP with no authentication and send authentication. Send most nearly works. Monitoring with pcapsipdump and wireshark shows inbound calls to FreePBX with RTP traffic working fine, but outbound calls to GXW4108 die after receiving a ‘501’ Not implemented msg to a REGISTER request. The userid and password that are sent from FreePBX are correct.
I’m about to pull my hair out…