[Asterisk-Users] Early Media with Asterisk

Hauke Zuehl hzuehl at athene.dnsalias.org
Thu Sep 22 03:26:37 MST 2005


Hi :)

I hope someone has a hint concerning Early Media.

The situation:
My Asterisk is connected to small local carrier who works with several SIP 
servers.
I traced some SIP headers and find something like this:
Via: SIP/2.0 UDP sip1.provider1.de

In the SDP part I found something like this:
o=- 2268929 0 IN IP4 sip2.provider1.de
c=IN IP4 sip2.provider1.de

If I send Dial(SIP/number|10|m(number)) I have silence on the line. No 
ringtone, nothing.

Now contacting a friend whose Asterisk is connected to another provider (let's 
give him domain provider2) traced this:
Via: SIP/2.0 UDP sip1.provider2.de

and its SDP looks like this:
o=- 2096205915 2096205915 IN IP4 sip1.provider2.de
c=IN IP4 sip1.provider2.de

and his early media works fine which means Dialing like the dial above works. 
The caller can listen to music :)

Btw: I wrote hostnames because it's an example. Originally there are IP 
addresses in SDP part.

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".

So I think Asterisk has a problem if another gateway than the original SIP 
server tries to connect. Is this correct?

Hope someone has a hint, but maybe it is an error in my provider's routing or 
configuration?

Thanks and kind regards,
Hauke



More information about the asterisk-users mailing list