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

[占平陈]

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 at 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?
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