Hello Michael,
i just tested your patch with my tcom setup. I noticed that it works in most
cases.
On case that leads to a fail is a reinvite because of codec or connect line
information change. Take a look:
Calls starts:
INVITE sip:0191...@tel.t-online.de SIP/2.0
Via: SIP/2.0/TLS
On 30.05.19 at 10:24 Michael Maier wrote:
> Hello!
>
> I wrote some code, which adds basic media encryption support to be used with
> Deutsche Telekom. The attached patch is based on Asterisk 16.3
> and works for me :-) - not fully tested yet. If you want to use it, you have
> to enable
