We just started to have a random one way audio issue with some of our clients outbound calls.
When the error occurs, the caller can hear the callie by the callie cannot hear the caller. If the caller tries again the call will work perfectly.
The odd issue is everything has worked for over a year now with no changes for over 3 months and the issue just started occurring.
Anyone have any suggestions on how to trouble shoot this issue?
FreePBX Distro 5.211.65-10 and 6.12.65-20
PBX systems sit behind a firewall with a static NAT (one to one NAT)
All RTP ports (10001 to 20000 UDP) are allowed for all sources
The phones are off site behind their own NATed firewall (some phone connect to VPN)
NAT settings are configured inside the “Asterisk Settings” setings
I’ve been letting this one sit on the back of my brain.
The closest I’ve gotten was a situation we had where the RTP ports were being walked on by our firewall.
- The firewall let some of the RTP traffic ports expire and the audio would just stop.
- The firewall wasn’t set up to forward the RTP ports to the server. I ended up changing the asterisk config to limit RTP ports to a couple per phone so that I didn’t have to leave the entire UDP address space open.
My experience with this stuff is that it’s almost always a firewall somewhere. Typically, it’s the one closest to the phone server, but it could be almost anywhere.
I have drilled down into my issue a little bit further and it appears our PBX is sending a cancel message within 2 seconds of the call starting which is causing our carrier to not accept any of the RTP packets from our PBX system.
Has anyone seen where the PBX will send a cancel message but still keep the call going?
Also what reasons (other than a hangup) would the PBX system send a cancel message?