The Asterisk dialplan application, Dial, offers the ability to add the number of seconds the phone can ring. It will wait for that amount of seconds (for example, 10) for the called party to answer and then move to the next priority. In this example we have the VoiceMail application. Please replace the /etc/asterisk/extensions.conf with this content:

[default] 
exten => _200[1-2],1,Dial(SIP/${EXTEN}, 10)
exten => _200[1-2],n,VoiceMail(${EXTEN},u)

Asterisk needs some additional configuration for the voicemail boxes. Please replace the file /etc/asterisk/voicemail.conf with:

 [general]
format = wav
attach = yes
[default]
2000 => 1234,Mr. X
2001 => 1234,Mr. Y

Now you can make a call and after approximately 10 seconds the Dial application stops calling. Asterisk increases the priority by 1 and starts the VoiceMail application for the voicemail box ${EXTEN}, and the calling party can leave a message.

Voicemail configuration in FreeSWITCH is very different. Please replace the file /usr/local/freeswitch/conf/dialplan/default/01_New.xml with this content:

<?xml version="1.0" encoding="utf-8"?>
<include>
  <context name="default">
    <extension name="Local_Extension">
      <condition field="destination_number"
       expression="^(200[1-2])$">
        <action application="export" 
          data="dialed_extension=$1"/>
        <action application="set" data="call_timeout=10"/>
        <action application="set" 
          data="hangup_after_bridge=true"/>
        <action application="set" 
          data="continue_on_fail=true"/>
        <action application="bridge" 
          data="user/${dialed_extension}@${domain_name}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" 
data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
  </context>
</include>

The FreeSWITCH dialplan is a bit more complex but also gives more control. With call_timeout=10 you can set the maximum time in seconds the bridge application tries to call the other party. The setting hangup_after_bridge=true tells FreeSWITCH to hang up after a bridged call has occurred. (This is also important for when the caller goes to voicemail and hangs up.) The setting continue_on_fail=true handles the scenario when the called party is busy. After the bridge application we have the answer, which means that FreeSWITCH kind of "picks up the phone" itself. After a one second sleep (which just feels a bit more human than without it) it bridges the call to a loopback destination for the voicemail application. You can also use the voicemail application without the loopback channel; however, by using this specific syntax we allow for attended transfers into a user's voicemail box.

Now that we've configured the system to record voicemail messages for our users, let's discuss how to access those messages. In each case, be sure to call and leave a voicemail message for a user and then follow the instructions for retrieving.

We'll create dialplans that allow us to check voicemail boxes by dialing 4000.

[default] 
exten => _200[1-2],1,Dial(SIP/${EXTEN}, 10)
exten => _200[1-2],n,VoiceMail(${EXTEN},u)
exten => 4000,1,VoiceMailMain(${CALLERID(num)})

<param name="password" value="1234"/>

<param name="password" value="1234"/>

    <extension name="vmain">
      <condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name}"/>
      </condition>
    </extension>

data="check default ${domain_name} ${caller_id_number}"