[solved] Thought it was SIP trunk issue, distro needed upgrading. one-way audio

FreePBX distro, 12.7.4-1804-2.sng7
Asterisk 15.

I am trying to get some SIP Trunks from Flowroute working. Currently everything works except inbound calls, the inbound audio does not work. Outbound audio does. But when I dial out, both inbound and outbound audio works.

Our FreePBX is assigned a publicly routable address, and I have verified theat the RTP UDP ports are open and the FreePBS is getting UDP packets on those ports during the call (verified using tcpdump).

Any ideas on what to check?

Appears the problem is not related to Flowroute or my SIP Trunks. It appears to be related to our Sangoma end point. I got a Yealink phone in (we’re currently evaluating Asterisk/FreePBX/Various hard phonoes).

Now, what appears to be happing, the Sangoma cannot receive audio, with the sole exception being if I call an external line from The Sangoma. In all other instances the Sangoma does not play any audio from the other end. Tested from soft phones, Yealink, and external lines.

Check the SIP ports the devices are using.

I have no idea why Sangoma set these defaults.

Because if you have more than one behind the same NAT you can certainly have weirdness if your router loses track of things.
This is my house with a Yealink T46G and a Sangoma s500.

This is a client with Yealink T42G everywhere. Not sure why the Yealinks are pulling from that low range. I’ll have to look into that. But you get the idea.

The Sangoma s500 is set to use 5060 as it’s local SIP port, and not to use a random port.

The FreePBX uses 5160 as it’s SIP port (5161 w/ TLS).

Currently the FreePBX and both phones are all on the same subnet. And the FreePBX has a publicly route-able IP address. So for now there is no NAT involved at all. I will work on and troubleshoot problems for NAT for remote hard phones when I get to it. For now I just need to figure out this audio issue.

I see that you’re using your Sangoma as a PJSIP endpoint. Do you recommend that I go that route? I currently have it set to chan_sip.

That is outbound and not the local port.

Yeah, so that means this is not your issues. Just tossing things out.

Ran tcpdump on the server, capturing UDP packets to and from the Sangoma, ports 10000-20000. I am seeing a ton of UDP packets from the phone to FreePBX, but only a couple packets from The Sangoma to FreePBX.

Also found the pcap feature in the s500’s web gui. Looking at that pcap inside wireshark reveals the same thing. Lots of SRTP packets going from the phone to FreePBX, but no SRTP or UDP going from FreePBX to the phone.

That’s usually an indication of a firewall configuration problem.

FreePBX and the s500 are on the same subnet, so there are no external firewalls or routers.

And I’ve tried turning off the firewall on FreePBX by doing a…

fwconsole firewall stop

Problem still existed.

However, I ssh’d to a shell prompt and did a yum update & reboot. That appears to have fixed the problem.

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