Secure_bridge_signaling and and secure_bridge_media

Asterisk
1

Hello,

I’m trying calls from BLINK SoftPhone to another SIP extension using TLS and SRTP

I’m using the two CHANNEL Options to see when the the calling channel use TLS and/or SRTP

The value of the two variables is always empty also when the call start from BLINK Softphone configured with TLS and SRTP. I can see the padlocks for TLS and SRTP on BLINK:

My dialplan:

exten => _100[0-2],1,Noop(Protocolo SIPTLS = ${CHANNEL(secure_bridge_signaling)})
same => n,Noop(Protocolo SRTP = ${CHANNEL(secure_bridge_media)})
same => n,Dial(SIP/${EXTEN},45,hHkKtTwWxX)
same => n,Hangup

Same behavior with PJSIP.

Any hint?

Regards

2

So is it a bug?
Regards