You need to enable SRTP on both sides, the PBX and the device. You need to show a call that fails as you say they are.
Encryption can be disabled, optional, or required. If one side has it disabled and the other required, the call will fail. In MicroSIP, each Account has a Media Encryption setting. In FreePBX, each extension has a Media Encryption setting, as well as an Allow Non-Encrypted Media (Opportunistic SRTP) option. The two ends must be set consistently.
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Obviously! And they are.
When media encryption in freePBX extensions are set to SRTP and on both sides of the call in MicroSIP it is set to None, the call fails as I’d shown above.
When in in freePBX extensions are set to SRTP and in MicroSIP it is set Optional SRTP, the call goes through normally but w/o encryption.
When in in freePBX extensions are set to SRTP and in MicroSIP it is set to Mandatory, the call fails s I’d shown above.
We haven’t seen a proper PCAP/sip debug of the failed call.
Most likely, there is no cipher in common between those that MicroSIP supports and those that Asterisk supports.
Unfortunately, my previous attempt to deal with this failed; see
If it also fails for you, I hope that an Asterisk wizard can chime in here.
As Asterisk/FreePBX is a b2bua and it is rare for your VSP to support encrypted media, what advantage do you envision?