In house direct_media=no does not work?

Asterisk Asterisk SIP
1

Newbie here.

(Using pjsip, and Asterisk 16.)

I’m having a lot of problems with a remote endpoint behind a NAT. But while trying to get that working, I became concerned about one thing.

I tested this thing and got results different than expected. If I configure the in-house telephone sets (Siemens OpenStage 15) for direct_media=no, then there is no more audio calling from one in-house phone to another.

Since I thought that meant that media would pass through Asterisk instead of direct, I expected to get audio.

If I take out that one line, then I get audio.

Am I misunderstanding something about direct_media, or is my Asterisk broken?

Thanks in advance!

2

Your configuration or environment is likely broken. You’d need to actually look at the SIP signaling (pjsip set logger on), the RTP traffic (rtp set debug on), and see how media is actually flowing.

3

Thank you very much!

Started digging around a bit, but then thought I’d try the same test on the server I first started building the Asterisk configuration on. Moved the configuration over, did the test, and worked fine.

So just going to rebuild the server it doesn’t work on.

4

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