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

Yungwei Chen yungwei at resolvity.com
Sat Jan 26 02:06:41 MSK 2013


I did a quick test against freeswitch- 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 (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" />

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+'');

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

I found one issue that voices are always out of sync in the recordings.
I am running freeswitch- 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"/>

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"/>

3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call.
        <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"/>

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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Michael S Collins
Twitter: @mercutioviz

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