[Freeswitch-users] Audio is not flowing , log is mod_sofia.c:1117 Asynchronous PTIME not supported, changing our end from 30 to 20

I use FS ,codec is ulaw ,when other softswitch call in,the audio is not flowing and distortion. the log is : http://pastebin.freeswitch.org/17517 I change my codec ulaw packet time to 30 ms (PCMU@30i) in external.xml,but the audio is not flowing also. I add <param name="rtp-autofix-timing" value="false"/> to external.xml, it no useful. Someone can tell me ,Why is it?