NO JOY One way audio using DIDWW

[totsuka]

I've got PBXF3 running on a Pi (I think it is the original Pi).

Anyway, it has been running for a few years no issues with several different VOIP providers.

Yesterday I registered for a DID from DIDWW as they have a lesser used country that I needed.

I have anonymous inbound SIP calls enabled (fun to watch the hackers in the logs when I allow 5060 to the world).

I pointed DIDWW to my server, no trunk setup. I configured an inbound route with the DID number and set the destination to the extension I want it to ring at.

Made a test call, extension rings.

When I answer I can hear the caller, but the caller can't hear anything.

Asterisk was doing RTP Allocating from port range 10000 -> 15000, but this DIDWW page says they need from 10000 - 30000. So I adjusted FreePBX via the GUI to extend the listening ports up to 30000.

Restarted to be sure, same result. One way audio. Here is a snippet of the Asterisk log, it actually thinks there is no RTP activity at all.

[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- Called SIP/8526
[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 is ringing
[2017-03-23 14:10:04] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 answered SIP/46.12.9.4-00000002
[2017-03-23 14:10:35] NOTICE[2991] chan_sip.c: Disconnecting call 'SIP/8526-00000003' for lack of RTP activity in 31 seconds


Any ideas where to look?

 

[totsuka]

Thought I would update this post, as I made a 'breakthrough'.

I may have actually had two problems, problem #1 may have been the port forwarding of RTP ranges not being large enough - causing the one way voice.

But turns out I created another problem for myself.

Years ago (2 or 3 years ago) our internal network was 192.168.1.0 - that was the local network we hat defined under the NAT/SIP settings in the GUI.

We changed our internal network to be 192.168.81.0 maybe 1 year ago. We never changed the local network information in the GUI, don't know how it could have been working all this time.

But this error that I was seeing:

Disconnecting call 'SIP/8526-00000003' for lack of RTP activity in 31 seconds

That was being caused by the incorrect local network value, once that was corrected the calls started working... Everything appears to be operational now including incoming calls from DIDWW.

 

[briankelly63]

Good catch. Consider Wireshark next time..