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 (http://wiki.freeswitch.org/wiki/Fs_logger.pl) comes very handy.

25 апреля 2012 г. 16:14 пользователь Andrew Paul <andrew.paul85@gmail.com> написал:
Hai,
I have a freeswitch setup  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 cli  switch_rtp.c "Auto changing IP PHONE IP to ASTERISK PBX IP " . 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 .

Thanks And Regards

Andrew Paul

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org