I want to force direct media over SIP trunk

I have tried:


But still the called phone tries to RTP directly to the calling phone when I want both SIP and RTP to go over the trunk. “core show version” on the CLI returns: Asterisk 11.13.0.

umm SIP has to stay on the PBX. You can send the RTP direct but not SIP

Let me explain a bit more. Phone A registers with asterisk A and calls Phone B which is registered with asterisk B. The two asterisk have a SIP trunk. In normal operations, SIP goes over the trunk for call setup and then RTP goes directly between the phones. I want to force RTP to also go over the trunk.

So when Phone B gets a call invite from Phone A, there needs to be some instruction that after the call is setup, RTP needs to be sent to asterisk B not Phone A.

If you setup a SIP trunk between the systems the SIP trunk will keep the RTP and SIP across that trunk. It wont be phone to phone.

I’ve done a packet capture on the B network and Phone B is sending RTP packets to Phone A’s IP address.

How is your SIP trunk setup and what options have you defined in the trunk