Setting up SPA3000 (No Sound when calling)

Managed to solve the previous registrar problem,

I’m trying to setup spa3000

I can now call from spa to pstn by dialing a prefix, but no sound is coming out.

“[2016-06-08 10:12:37] NOTICE[5496] chan_sip.c: Disconnecting call ‘SIP/pstn-0000000f’ for lack of RTP activity in 31 seconds” I believed this is the error message.

May I know if anyone has the experience in setting up this device?

FreePBX 13.0.119
SPA3000 SW 3.1.20(GW)

Did you check the wiki? (I did :wink: )

yes i did follow the wiki instructions.

my spa3000 couldn’t get registered sometimes
"NOTICE[29207] res_pjsip/pjsip_distributor.c: Request from ‘“pstn” sip:[email protected]’ failed for ‘’ (callid: [email protected]) - No matching endpoint found"

when I have already pointing the correct port number for chan sip, it will still go to the wrong pjsip registrar

What in your opinion is the “correct” port number for chan_sip on your server, does the ATA agree? (It looks like you got it backwards because the ATA is actually talking to pj_sip)

I have my Chan_SIP bind to port 5061 (the default port), as for the ATA part I have set the port to 5061 as well,
is there anything that I actually missed out?

If i were to use the pjsip as trunk, it can be registered successfully but will always complaint that the trunk is congested when i’m trying to make a call out.

Sorry to trouble you as I’m quite new to this.


This is a guy in Malaysia trying to connect to your pj-sip service. So probably spurious but you need to crank up your firewall/IDS is apparently an “Own Cloud” instance, check that it allows UDP/5061 through to and yuour NATing is correct

Not sure if i’m making it too complicated.

I have a raspberryPi running an owncloud server as the DMZ, while there’s a freePBX server running behind it which I have the done the port forwarding of 5060,5061 and RTP ports.

Now I’m trying to connect SPA3000 from Malaysia to the server in Singapore.

Sometimes the pstn registered successfully, but there’s no voice as the RTP lack of activity problem comes out again.

Thanks for your help!

With respect you are way overstretching a Raspberry Pi, Just start off slow with a basic RasPBX and ask in

or another route

thanks! just one last question

Do you know why is my voip call mapped rtp port being mapped to port 0 in this situation.

Sorry , no. Whatever did that should not do it, it won’t work :slight_smile:

I can assure you thought that it wasn’t Asterisk nor FreePBX, it was one of your higher/lower layers (depending on how you look at it)