[Asterisk-Users] Early Media with Asterisk

[Hauke Zuehl]

Hi :)

Am Donnerstag, 22. September 2005 12:48 schrieb Andreas Sikkema:
> asterisk-users-bounces at lists.digium.com wrote:
> > Now, I traced RTP packets and see how sip2.provider1.de sends
> > packets to my Asterisk but the port seems closed on my server so the
> > inquiring server of
> > provider1 will never get an answer and sends a "port unreachable".
>
> Did provider1 send the exact same SIP message types to you
> as provider2? It looks to me like provider1 is not sending
> a 183 Session Progress message. Which is usually used for
> this kind of functionality I think.

Oops!
Mistake by myself:
They start with INVITE (sure!) with all the SDP stuff. I answer 100 followed 
by 183.
So I do send the 183 message to provider1 and provider2.

Oh, I've forgotten to tell: My problem are incoming calls with early media  
from PSTN to my server.

Sorry for that :)

Regards,
Hauke