On 21.10.19 at 17:23 Michael Maier wrote:
Attached are actual patches for Asterisk 16.16.0-rc1, 18.0.1 and 18.2.0-rc1 (one patch for each version). Version 16.16.0-rc1 is only compile tested (16.14.x was the last asterisk 16 version I used myself). 18.2.0-rc1 is used here (as 18.1 and 18.0.x, too).
All three patches contain a fix for the handling of 183 Session Progress w/o SDP, which is handled as 180 Ringing. Another fix handles 180 Ringing w/ SDP as 183 Session Progress. You may remove them if you don't like them. They are not mediasec related. But they reflect "features" of Deutsche Telekom to be sure to get ring back and early media for my C610IP.
How should it all be used now? If you want to use SIPS and SRTP with Deutsche Telekom AllIP, you have to be sure to enable the following features in the pjsip trunk (endpoint): - transport: tls (TLS 1.2) - enable SRTP for this trunk - endpoint: support_mediasec=1 - registration: support_mediasec=1 If you are using FreePBX, you have to add the support_mediasec switches to pjsip.endpoint_custom_post.conf and pjsip.registration_custom_post.conf. This is done like this: File pjsip.endpoint_custom_post.conf: [your name of the trunk](+type=endpoint) support_mediasec=1 File pjsip.registration_custom_post.conf: [your name of the trunk](+type=registration) support_mediasec=true
Thanks Michael
mediasec.tar.xz
Description: application/xz
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
