[asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters

[Michael Maier]

Hello!

That's the *complete* patch for Asterisk 16.3 (can be applied to asterisk 16.4, too). Logging has been changed to ast_debug(3,).

See previous posts for information about usage.


Regards,
Michael
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