RTP learning with no sound, until DTMF is pressed

FreePBX Endpoints
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1

Hi community,

I have noticed a very strange issue in the following scenario.

Environment:

  • Freepbx 16
  • Asterisk 18.16.0
  • Installation on cloud VPS (Real ip reachability)
  • Trusted SSL Certificate from let’s encrypt system wide

Device:

  • IP Phone Yealink T31P

Configuration:

  • Calls are sent out or come thought a sip trunk
  • Extension registered on the phone using TLS

Issue:
The registration works perfectly, after setting up the configuration on the phone, account, extension, password, domain/ip … etc.
After initiating a call to a remote party through a SIP Trunk, or receive an inbound call to the yealink phone, the channel starts fine, with no sound going out from my side on the phone, until I press any digit as DTMF, right after that, I get that asterisk message:
asterisk Strict RTP switching source address to x.x.x.x
and my voice starts to be heard immediately !

any ideas ?

2

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