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

[Michael Maier]

On 21.10.19 at 17:23 Michael Maier wrote:

New patchset for Asterisk 18.4. As I don't use other versions of Asterisk any more, I don't have a patchset for those versions.

> 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
> Regards
> Michael

-------------- next part --------------
A non-text attachment was scrubbed...
Name: mediasec-18.4.tar.gz
Type: application/gzip
Size: 4810 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20210515/faaf8082/attachment.gz>