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
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