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

Wed Jan 30 11:42:08 MSK 2013

Did you try other voices?

Are you sure you have built the latest version of flite?

Download and build flite again.

I'm pretty sure this was fixed some time ago.

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Monday, January 28, 2013 23:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voices in the recordings are out of sync

Ok... You should be testing against MASTER... If you are not testing again
Master It is not going to get fixed... All patches go there first... If you
can duplicate there, then open a ticket. This is the only way these sort of
things get fixed

On 1/28/13 4:43 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

Tested against the following version today, and the problem is still there. 
FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765
2013-01-04 15:45:59Z)

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei
Chen
Sent: Friday, January 25, 2013 5:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voices in the recordings are out of sync

Hi,

I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem
is still there.
In the recording, the caller speaks way faster than he did.

Here's the version of freeswitch:
FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7
2012-12-29 00:17:59Z)

Here's the set of steps:
1. In dialplan/default/resolvity.xml, add the following extension:
  <extension name="ext1">
      <condition field="${destination_number}" expression="^0009$">
        <action application="set" data="RECORD_STEREO=false"/>
        <action application="answer" />
        <action application="javascript" data="test.js" />
      </condition>
  </extension>

2. Create scripts/test.js with the following content. This js file will read
numbers sequentially starting from 0.
       session.execute("record_session", "/tmp/test.wav");
       for (var i=0; i<100; i++)
       {
            session.speak("flite", "kal", i+'');
       }
       session.hangup(16);

3. reloadxml
4. dial 0009 from a registered SIP client, and then repeat each number you
heard.
5. Listen to the recording, /tmp/test.wav.

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 18, 2012 8:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voices in the recordings are out of sync

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.

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-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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