[Freeswitch-users] bad audio of recorded session

[Piotr Sobolewski]

On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> It is a known-issue that you never know what you're going to get with
> Asterisk.. yes.

Is there a way to solve this issue without taking away asterisk?

>
> 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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-- 
Piotr Sobolewski
sobolewski at gmail.com