[Freeswitch-users] record_session wav files are corrupted

[Ian de Sousa]

Hi All,

Just wondering if there has been a fix for this issue.

Im running 1.4.19 and occasionally we are getting calls playing very quickly when played back.. then the last bit of the recording / call it switches to normal speed at about 1/2 way.

testing so far (100 calls) seems to indicate unless we set an “answer” beforehand we get this issue 50% of the time.
setting answer isn’t good as caller will get incoming call charge due to specific NGN numbers coming in.


Ian DS
#######

Do you have mod_sndfile loaded?

Steve

On 4 Apr 2013, at 16:06, fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> wrote:

> On 04-02-2013 05:17, Steven Schoch wrote:
>> On Tue, Apr 2, 2013 at 10:28 AM, <fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>> 
>>>  If I try to import the
>>> audio as raw format in audacity you can hear some high pitched 
>>> talking
>>> that plays back very quickly.
>> 
>> When you import as raw data, try changing the sample rate to 8000.
>> 
>> The reason Audacity can't open the file is because the header, which
>> specifies the encoding, channels, sample rate, etc. is damaged or
>> missing.
>> 
>> -- 
>> Steve 
>> _________________________________________________________________________
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>> 
>> 
>>  <http://lists.freeswitch.org/pipermail/freeswitch-users/2013-April/094369.html>
>> 
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> 
> Well why would the header to damaged or missing. I have updated the 
> config to the following and it still doesnt work. If I look at the 
> header info with windows by going to file properties it doesnt list 
> anything in the header info. The file is just a bunch of high pitched 
> noise if I dont import as raw. I can hear the audio fine when importing 
> raw and 8000hz rate. I am not sure what else I need to do to make this 
> work.
> 
> I followed the steps listed on: 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session>
> 
> <extension name="bandwidth.10d" >
>    <condition field="destination_number" expression="^(\d{10})$" >
>        <action application="set" 
> data="sip_h_X-accountcode=${accountcode}" />
>        <action application="set" data="sip_h_X-Tag=" />
>        <action application="set" data="call_direction=outbound" />
>        <action application="set" data="hangup_after_bridge=true" />
>        <action application="set" 
> data="effective_caller_id_name=${outbound_caller_id_name}" />
>        <action application="set" 
> data="effective_caller_id_number=${outbound_caller_id_number}" />
>        <action application="set" data="inherit_codec=true" />
>        <action application="set" data="record_sample_rate=8000" />
>        <action application="set" data="RECORD_TITLE=Recording 
> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}" 
> />
>        <action application="set" data="RECORD_SOFTWARE=FreeSWITCH" />
>        <action application="set" data="RECORD_STEREO=true" />
>        <action application="set" data="RECORD_DATE=${strftime(%m-%d-%Y 
> %H:%M)}" />
>        <action application="set" data="RECORD_ARTIST=FreeSWITCH" />
>        <action application="set" data="RECORD_COPYRIGHT=(c) 2013" />
>        <action application="set" data="RECORD_COMMENT=FreeSWITCH" />
>        <action application="record_session" 
> data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" 
> />
>        <action application="bridge" 
> data="sofia/gateway/customer1-bandwidth/$1" />
>    </condition>
> </extension>
> 

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