RTP Transmission error of packet to 0.0.0.0:5014

Asterisk Asterisk Support
1

I’ve set up a new ATA (Grandstream HT503) and configured it to handle my PSTN/hardphone.

However, whenever there are communitacoins involving either line, there seems to be no outgoing sound, and the terminal is flooded with lines that read:

The full log is here. Here’s a brief summary of the extensions so it’s understandable:

778: The extension called by the ATA when there’s an incoming call on the PSTN.
777: The user used by the ATA for the PSTN line.
700: The user used by the ATA for the phone line.
701, 702: Laptop/desktop

The issue is reproduced by any call involving the ATA (ie: call from mobile phone to the my pstn number, and answer from the laptop, call from the hardphone to an external number, etc).

Any ideas? :frowning:

2

0.0.0.0 in the SDP should be treated as a hold.

You need to enable sip debugging, and turn up the debug and verbosity levels (5 is the value suggested when creating bug reports) so that one can see what SDP was received and how Asterisk handled it.