[Freeswitch-users] voices in the recordings are out of sync

[Michael Collins]

Latest version of FreeSWITCH has some updates that may fix this issue. I
would update to 1.2.5.3 ASAP.
-MC

On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I found one issue that voices are always out of sync in the recordings.
> I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed
> from yum.
> I am having trouble installing the latest version from source due to an
> error: Autoconf version 2.62 or higher is required.
> It would be nice if someone can reproduce this issue against HEAD. Thanks.
>
> Here're the steps to reproduce it. The idea is to call a phone number and
> then bridge to another phone number while the entire session is being
> recorded.
> 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10
> digit phone numbers.
>     <extension name="public_extensions">
>       <condition field="destination_number" expression="^\d{10}$">
>         <action application="transfer" data="main XML default"/>
>       </condition>
>     </extension>
>
> 2. In dialplan/default/main.xml, make sure you have an extension to handle
> the call in the default context.
>   <extension name="test">
>       <condition field="${destination_number}" expression="^main$">
>         <action application="set" data="RECORD_STEREO=false"/>
>         <action application="answer" />
>         <action application="record_session"
> data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>         <action application="bridge"
> data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/>
>       </condition>
>   </extension>
>
> 3. In sip_profiles/external/gateways.xml, make sure you have a gateway
> that allows you to make an outbound call.
>     <include>
>         <gateway name="gw1">
>           <param name="username" value=""/>
>           <param name="password" value=""/>
>           <param name="realm" value=""/>
>           <param name="from-domain" value=""/>
>           <param name="extension" value=""/>
>           <param name="expire-seconds" value="60"/>
>           <param name="register" value="false"/>
>           <param name="retry-seconds" value="60"/>
>         </gateway>
>     </include>
>
> 4. make a call to one of the allowed 10-digit phone numbers in your
> environment.
> 5. Once the call is answered, the caller shall start to count from 1 to 60
> with some pause after each number.
> 6. The callee shall repeat each number he/she heard from the caller.
> 7. You should be able to hear that 2 voices in the recoridng
> (/tmp/rec.wav) are out of sync.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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