Good day everyone,
I’ve got a couple of issues (3 part question) with one of my Labs (PJSIP configuration) :
I have two labs and have deployed servers with mirroring PJSIP configurations; however my results are night v day.
Problem 1: No audio between local PJSIP endpoints A & B connected to PBX1. ( SIP signaling using port 5160 / method=UDP ) - using port 5160 to prevent conflicts with working platform that uses port 5060.
In this scenario, I grabbed packets from the server and see the SIP signaling passed and processed as normal ( Endpoint A to PBX1 to Endpoint B & vice versa); however, when audio is generated, I see RTP packets reaching the PBX1 interface from the source endpoint (A), but is not forwarded to the remote endpoint (B) and vice versa ( DIDs ie. 2000 and 2001) . I was able to replicate these results on a second serve in Lab 1. ( I see the RTP be relayed to the correct negotiated IPs and Ports within SDP but the server fails to relay the Audio to the 2nd leg of the call)
however, in my Home Lab (Lab 2) the replicated configurations and scenarios work like a champ.
it seemed almost similar to an issue ( below), reported a cple of weeks ago, but not apples v apples - does anyone have any suggestions offhand? -> can forward configs l8r
"# [No Audio on Public-facing FreePBX 15 and Asterisk 16 on CentOS 7] "
Scenario 2 : PJSIP trunk registrations fail over the internet but the Trunks remain online
2nd issue : PJSIP trunk registrations are rejected with 403’s but remain online and work ( atleast for my home lab) Is this expected behavior for PJSIP trunks? Also what PJSIP trunk configurations would be deemed the most secure over the internet?