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

[Michael Maier]

Attached you find a new patchset for Asterisk 18.11.

Thanks
Michael

On 15.05.21 at 08:15 Michael Maier wrote:
> 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
> 
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