> Voice Mail and Asterisk PBX

Voice Mail and Asterisk PBX
As the name implies, the voice-mail feature provides capability for callers to leave
messages when the called party is unavailable. Asterisk PBX supports a local voicemail
(VM) system that offers many options, including password protection, system
greetings, e-mail notifications, and VM forwarding.
Asterisk’s voicemail.conf serves as the main VM configuration file that defines
voice-mail boxes and their options.This voice-mail configuration file is, by default,
located in the /etc/asterisk directory.The syntax for creating a voice-mail box inside
voicemail.conf file is as follows:
mailbox_num =>
voicemail_password,user_name,email_address,pager_email_address,option(s)
mailbox_num defines the number of the mailbox extension
voicemail_password defines the user’s password to access VM options and
retrieve voice messages
email_address defines the e-mail address where the new VM notifications
can be sent, including the audio of the VM
pager_address defines the e-mail address where notifications of a new VM
can be sent
options define additional available VM options that the user can set
Figure 2.17 displays an example of a mailbox configuration in the voicemail.conf
file. In this example, mailbox 2001 and 3001 for Softphone2 and Softphone3 were
created. Mailbox 2001 was configured with password 123456 and the e-mail address
of.
Figure 2.17 Configuring voicemail.conf Configuration File
Now that we have configured mailboxes 2001 for Softphone2 and 3001 for
Softphone3 users, we must configure our dial plan to allow callers the option of
leaving voice mail for these two extensions.To add voice-mail capability in our dial
plan, we need to edit our extensions.conf file with the VoiceMail() function to extensions
2001 and 3001. Figure 2.18 displays part of the extensions.conf file and needed
configuration.The argument inside the VoiceMail() function refers to the
mailbox_num@context, which in this case is 2001@default for Softphone2 and
3001@default for Softphone3.
Now that we have created our password protected mailboxes and modified our
dial plan to allow callers to leave messages, we must designate an extension number
that local users can dial to gain access to and retrieve their voice messages.We
accomplish this task by adding the VoiceMailMain() function to our dial plan and
assigning the extension *98 that we have picked for our case study.The line added
to the extensions.conf file will look like this: exten => *98,3,VoiceMailMain(default).
Now callers can leave VM and end-users can retrieve them by dialing *98 and
their mailbox number. Figure 2.19 shows the log file from Asterisk PBX of an actual
call in progress where the caller is leaving a VM for extension 3001. Notice that the
VM is saved as wav and wav49 format audio file under the /var/spool/asterisk/voicemail/
default/3001/INBOX/ directory.



How Is VoIP Different
from Private Telephone Networks?
VoIP technology brings many new capabilities to today’s modern communications.
The biggest difference between VoIP and traditional private telephone company networks
is the transport protocol. Internet Protocol, which we identify as the transport
protocol that we use for services such as viewing Web sites, playing online games,
and sharing music, now is being used to carry our voice.VoIP is truly a revolutionary
communication technology with rich new functionality that is being
accepted by communities around the world.



Circuit-Switched and
Packet-Routed Networks Compared
Packet-routed networks (such as IP networks) are designed to move data. Data is
moved from source IP address to destination IP address in packets. Data from the
source is dynamically broken down and encapsulated into packets before it is sent to
its destination. Once the packets reach the desired destination, they are reassembled
into their original data format.The Internet is constructed mainly based on packet
routed technology, although many WAN links remain connected to circuit-switched
networks such as Asynchronous Transfer Mode (ATM).
Circuit-switched networks (like the PSTN) are designed to move voice or data.
Telephone networks that carry your traditional phone calls are circuit based.A
point-to-point dedicated static connection in a circuit switched network is required
prior to sending voice or data signals across multiple switches inside PSTN. Once a
point-to-point static channel is allocated and nailed up in the circuit switched network,
it will belong only to that one voice call and no other network traffic.
Packet-routed and switch networks both carry data, video, and voice. Packetrouted
networks carry voice on top of IP. By breaking the voice, data, or video into
small packets inside a packet-routed network, each packet can travel throughout the
network across multiple independent paths and later be reassembled at its final destination.
In circuit-switched networks voice and data travel across statically nailed-up
paths. Both circuit and packet networks have the ability to carry different traffic
across a network.The benefit of a packet-routed network is its dynamic approach
and ability to self-recover with failed link paths because packets do not have statically
allocated paths. On the other hand, the benefit of a circuit switched network is
its dedicated predefined path, which assures quality for that session by not sharing
with other resources.
Resource Reservation Protocol (RSVP) runs over IP and is used in packetrouted
networks. RSVP is a type of QoS that signals across a packet network and
allocates resources prior to making a voice call. RSVP applies circuit-switching principles
to packet-routed networks but even the best RSVP guarantee cannot match
the timeslot reservation guarantee inherent in circuit switched networks where a
statically defined path is used by only one resource.