[Freeswitch-users] One way Audio

[Anton Kvashenkin]

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 at 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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