[Freeswitch-users] bad audio of recorded session

[Anthony Minessale]

It is a known-issue that you never know what you're going to get with
Asterisk.. yes.


On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> Hi
>
> When conversation of the call is being saved to a file
> (record_session) audio tempo in such recording is like slowed down
> despite that call conversations lacks this issue.
> Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
> If the call comes from IP-Phone, recording is normal.  (UA-->FS-->Asterisk) OK
>
> I've tried multiple options for example:
>  <action application="export" data="rtp_rewrite_timestamps=true"/>
>  <action application="set" data="bridge_generate_comfort_noise=true"/>
>
> And setting  <param name="rtp-timer-name" value="none"/> in the sofia
> profiles didn't solved the problem either.
>
> Global Codecs definition.
>
>  <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>
>
> Is this known issue?
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
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-- 
Anthony Minessale II

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