You are here

No audio on calls

10 posts / 0 new
Last post
No audio on calls

Hi guys, apologies if this has been posted before but I reviewed several threads and didn't find anything.

I've setup FreePBX on my raspberry pi connected to my node via my switch. Three pjsip extensions were created properly (101 is mine, 102 is my fathers KP4RGD and 103 another fellow hams). I tested first with Linphone, entered the SIP server settings and it was able to register fine on the PBX. Im connected to my fathers node via Tunnel, provided him with the SIP server and extension details. He's able to call me, it rings and able to pickup, but we get no audio. Same from within my side. I setup a Cisco SPA 502G which registered fine as well. Tried calls from there to my fathers extension and the same issue happens no audio. I went ahead and opened ports on my home router for 5060 and did so on my node as well (Dad did the same thing) and it still didn't work. Anything im missing here? I ran a tcpdump and its showing good as far as I can read. The goal here is to set this up as an example for the AREDN team in Puerto Rico setting their nodes up. 

Thanks in advance!
Raul G. KN4FMO

nc8q's picture
it rings and able to pickup, but we get no audio.

Maybe no matching CODECs?
I am running a rasPBX and am successful with registering and calling Grandstream GXP1630/1620 and Nortel 1535 phones.
Also linphone and GSWave softphones.

In Applications->Extensions[edit the extension with no audio] Advanced[Edit Extension] Ensure that there are no 'Disallowed Codecs'.
In Settings->Advanced Sip Settings->General SIP Settings[Audio Codecs] I have the ulaw, alaw, gsm, and g726 boxes enabled.

I hope this helps, Chuck

Thank you for your quick

Thank you for your quick response Chuck, I indeed had several codecs unchecked so I just checked them all on just in case but nada. Im starting to suspect NAT issues, the most important test (testing locally withing my LAN) I havent performed yet, so ill borrow a phone and see if it works. 

KD1HA's picture
I remember this issue about 4

I remember this issue about 4 years ago with my Raspberry Pi PBX. I had to set the node with the PBX to port forward to 10000-20000 UDP in the Outside Port setup page with the IP address of the PBX. I'M not sure about the LAN port but I think I set it to the same?!?

Denis KD1HA

Thanks Denis! If you ever

Thanks Denis! If you ever find the LAN port you use for sure let me know, I can setup a range on the Outside port but not on the LAN port box.

KD1HA's picture
I can't confirm but Willis

I can't confirm but Willis KB1JFG thinks that we left LAN port blank and we set it up on both our ends. Give it a try and let us know the outcome.

kc8ufv's picture
If these are all on the mesh,

If these are all on the mesh, double check that the nodes in question all are set to x Hosts Direct, where x is at least as many devices (computers, phones, cameras...) are used on the local network. SIP sometimes has issues in NAT is used. Next thing to check would be if the phones both work when attached to the same node as the PBX. If they do, you'll want to run a long ping from a computer on the remote node to the PBX server. Ideally you'll have 0 packet loss, and the minimum and maximum latency will differ by only a few ms. If you're seeing more than a couple percent loss, or upper tens of ms or higher variance in latency (jitter), your link is likely to be unable to sustain a realtime audio connection.

Check SIP Settings


I had a similar problem when I replaced my 2 year old RasPBX with the latest version. I think old version defaulted to SIP, new version defaults to PJSIP, which has a lot more settings. In the old version, I simply defined extensions and everything worked out of the box. With the latest version, after I defined the extensions I could place a call between extensions, but had no audio.

The extensions and RasPBX are all on the same LAN subnet with DHCP reservations, so not a NAT or routing issue.

I ran a Wireshark trace and found that the SIP packets flowed to set up the call between extensions, but when call was answered RasPBX told the phones to use an external address to route the RTP audio packets.

Go to 'Settings/Asterisk SIP Settings/General SIP Settings' and look under 'NAT Settings' for 'External Address' - it is probably 88.x.x.x.
Change that address to the IP of your RasPBX and try again.

That works for me because all my extensions and RasPBX are on the mesh.
It is more complicated if you have devices on the WAN.
Someone else may be able to provide assistance on how to properly set up NATing for that use case.

John K2QA

Thank you for the info John!

Thank you for the info John! I checked and indeed it was set to my WAN "External Address" I changed it to my RasPBX IP but didnt work. I suspect its a NAT/Port issue that I haven't figured out since im connecting to the other node via WAN. Ill keep on working on it though!



I was able to resolve the issue, after extensive research and port forwarding, I linked my node via tunnel directly to my fathers and after doing so we were able to get audio. Looks like something going on with the network on the original node I was running through tunnel. Thanks to everyone that pitched in!

Theme by Danetsoft and Danang Probo Sayekti inspired by Maksimer