I have a signaling issue with PJSIP. Here is my setup:
Flowroute DID → Asterisk 16.17.0 Server → ATA registered to Asterisk
I’m using PJSIP on both legs of the call (I have a Flowroute endpoint and an ATA endpoint in pjsip.conf)
If I call the DID from my cell phone and then I hang up the cell phone, neither the Asterisk server nor the ATA sees the hangup. I don’t have the issue with chan_sip but I do have the issue with I use chan_pjsip.
It took me a while, but I’m pretty sure I tracked the issue down. I’m doing direct media, and as expected, when the ATA answers Asterisk sends a re-invite and then the media goes direct. However, the re-invite is switching the transport from port 5062/udp to port 5061/TLS.
I can’t use TLS in this scenario, but I can’t figure out how to stop that from happening. Here’s what I’ve tried:
- I created a new transport for port 5062 tcp, but that didn’t help.
- I tried setting disable_tcp_switch=yes in pjsip.conf but that didn’t help either.
- I tried setting symmetric_transport=yes on the UDP port 5062 transport, but that didn’t help.
Can someone please tell me how I can either keep everything on port 5062/udp or at least avoid using TLS? I use TLS with Flowroute for other purposes (hence why I have a TLS transport) but I just can’t use it here.
I’ve uploaded some scrubbed logs.
tls switch.txt (5.7 KB)