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

Michael Maier m1278468 at mailbox.org
Mon Jun 10 09:01:17 CDT 2019


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.

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