[asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1)

["Hans-Jürgen Brand"]

Asterisk version 1.2.14

I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.

Any hits for me?

*CLI> rtp debug
RTP Debugging Enabled
    -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
    -- Called snom
    -- SIP/snom-00797110 is ringing
    -- SIP/snom-00797110 is ringing
    -- SIP/snom-00797110 answered SIP/xlite-007918f0
    -- Attempting native bridge of SIP/xlite-007918f0 and SIP/snom-00797110
Got RTP packet from 192.168.100.70:50002 (type 0, seq 6022, ts 32652224, len 160)
Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49874, ts 64, len 160)
Got RTP packet from 192.168.100.20:17548 (type 0, seq 6911, ts 1973300, len 160)Sent RTP packet to 192.168.100.70:50002 (type 0, seq 28956, ts 16, len 160)
Got RTP packet from 192.168.100.70:50002 (type 0, seq 6023, ts 32652544, len 160)
Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49875, ts 384, len 160)       




*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
snom/snom                  192.168.100.70   D          2051     Unmonitored
xlite/xlite                192.168.100.20   D          11420    Unmonitored
2 sip peers [2 online , 0 offline]