<div class="gmail_extra">Without any siptraces and logs or dialplan snippets it would be extremely difficult to debug this one-way audio. Please, post this one. For that purpose fs_logger (<a href="http://wiki.freeswitch.org/wiki/Fs_logger.pl">http://wiki.freeswitch.org/wiki/Fs_logger.pl</a>) comes very handy.<br>
<br><div class="gmail_quote">25 ÁÐÒÅÌÑ 2012šÇ. 16:14 ÐÏÌØÚÏ×ÁÔÅÌØ Andrew Paul <span dir="ltr"><<a href="mailto:andrew.paul85@gmail.com" target="_blank">andrew.paul85@gmail.com</a>></span> ÎÁÐÉÓÁÌ:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hai,<br>I have a freeswitch setupš in that one asterisk pbx registered.
For this setup am facing problem with lower version of yealink phone and
grandstreams . Whenever 200 ok came in freeswitch changing IP phone ip
to asterisk PBX ip.It is showing messages in FS cliš switch_rtp.c "<b><i>Auto
changing IP PHONE IP to ASTERISK PBX IP </i></b>" . So no audio is passing to
PBX from from IP phone. Still IP phone able to get the incoming calls.
Both PBX and IP are in same network. Hopping the earliest reply .<br><br>Thanks And Regards <br><span class="HOEnZb"><font color="#888888"><br>Andrew Paul<br>
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