No audio between local extensions


I have a fresh RasPBX install and I am able to both receive and make external calls through trunk as well as make calls between extensions. The connection works but there is no audio/media on either of these calls. I have one DECT phone (ATA) and one soft-phone (Groundwire on iOS), all on the same subnet as the PBX.

The install is currently set-up to be ran with local extensions, but inbound and outbound calls are being made through respective trunk providers. I believe my NAT Settings (Local and External Addresses) are configured correctly and I’ve restarted Asterisk afterwards.

I hear my own voice when making a test echo call from an extension as expected, but between two extensions or external call there is no audio at all. I have read that I need to port forward ports 10-20k to the PBX, so that’s what I did. (Do I need to do this since I’m running locally?)

Test call with no audio between two extensions (pjsip log):

Happy to hear back from you guys!

No audio usually means no RTP between the endpoints. Are you certain all the required ports are open on the network?

I only have ports 10000-20000 forwarded on UDP to the PBX from any IP. As of now, I don’t want to open up port 5060 so I use only local extensions. This should be enough right?

rtp set debug on

from the asterisk cli will normally identify rtp connections’ traffic they (the put and get’s) would need to be symmetric and continuous for audio to be heard

Just called from my DECT phone to my iOS extension, the DECT extension is running via my ISP modem and calling “from” my public ip, thus I’ve redacted this in the log.

Dumped the RTP log here (redacted :

Asterisk is a B2BUA , thus two call flows would be expected unless direct media was requested and honored, what is ?

Not really sure what you meant by that, but happens to be my iOS extension.

What was the other party that was trying to talk to?

The other party is the DECT extension (router as ata adapter) which calls “from” my public IP. There is no further configuration available on the router.

What is the address of the PBX, there should be a flow from either endpoint to and from it.

The address of the PBX is

Good to know, a echo call works fine

make sure you have direct media disabled first

I just tried setting direct media on NO on both extensions and restarting Asterisk afterwards, but this didn’t resolve the issue. Is this what you meant?

Now look at the SIP session to see what SDP session was negociated. sngrep or wireshark are useful tools


[ ] 1    INVITE     [email protected]:5060       [email protected]:5060       11    REDACTED_PUBLIC_IPV4:5060      COMPLETED
[ ] 2    INVITE     [email protected]            [email protected]             7       COMPLETED

Look inside the INVITE for the negociation.

Just did as you said, I’m new to this. Hope you can spot the issue.

Now look inside the other INVITE

These are the contents of both invites, other invite starting at line 35.