El jue., 28 de feb. de 2019 a la(s) 09:21, Daniel Tryba (d.tryba@pocos.nl) escribió:

You wrote something else in your original message:
exposing asterisk RTP 10000-30000 makes it function.

that's the problem here! yes!

I read that to be an indication that rtp(proxy|engine) isn't rewriting.

in fact, the first try was with rtpproxy, but due the DEBUG log are very poor

we tried with rtpengine, and now i noted that rtpengine only output in log that:
```
Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28721]: DEBUG: timer run time = 0.000005 sec
Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28803]: [1551364784.000249] DEBUG: timer run time = 0.000035 sec
Feb 28 14:39:45 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000010 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000007 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28732]: DEBUG: timer run time = 0.000008 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28832]: DEBUG: timer run time = 0.000005 sec
```

BUT: rtpproxy seems are working cos if not there's no SIP sesion stablished!

but as i siad, no sound if i not exposed the asterisk ports!

NOTE: audio of the call are udp! so that means are not possible? i guess must be possible

due the kamailio can manage an route using the rtp[proxy|engine] right?

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