[asterisk-users] switching from simple_bridge technology to native_rtp issue

[Sameer Rathod]

Hi,

with canreinvite=no and directmedia=no I and getting the message in the
logs for all calls

"switching from simple_bridge technology to native_rtp"


-- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/102
    -- SIP/102-00000018 is ringing
    -- SIP/102-00000018 answered SIP/101-00000017
    -- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
    -- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
       > Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
simple_bridge technology to native_rtp
       > 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
       > 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
       > 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
       > 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
    -- Channel SIP/101-00000017 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
    -- Channel SIP/102-00000018 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
  == Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'



I cannot understand why asterisk state diff bridges if all works same

please can anyone explain me the working bridging concept and how to
configure and use bridges to route the rtp externally form asterisk.

-- 
Regards
Sameer Rathod
8109413462
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