FreeSWITCH 1.0.6: SIP and the User Directory

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Build robust high-performance telephony systems using FreeSWITCH

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by Anthony Minessale Darren Schreiber Michael S. Collins | August 2010 | Networking & Telephony Open Source

In this article, by Anthony Minessale, Michael S. Collins and Darren Schreiber authors of FreeSWITCH 1.0.6, we will learn about how we use SIP to connect users, both locally and around the world. SIP is a ubiquitous protocol in the VoIP landscape. In this article, we will:

  • Learn the principle behind the FreeSWITCH user Directory
  • Explore and configure the FreeSWITCH user Directory for the first time
  • Learn how to connect FreeSWITCH to service providers
  • Make modifications to the Dialplan and directory XML configuration
  • Briefly discuss SIP profiles and User Agents

(For more resources on Telephony, see here.)

Understanding the FreeSWITCH user directory

The FreeSWITCH user directory is based on a centralized XML document, comprised of one or more <domain> elements. Each <domain> can contain either <user> elements or <groups> elements. A <groups> element contains one or more <groups> elements, each of which contains one or more <user> elements. A small, simple example would look like the following:

<section name="directory">
<domain name="example.com">
<groups>
<group name="default">
<user id="1001">
<params>
<param name="password" value="1234"/>
</params>
</user>
</group>
</groups>
</domain>
</section>

Some more basic configurations may not have a need to organize the users in groups so it is possible to omit the <groups> element completely, and just insert several <user> elements into the top <domain> element.

The important thing is that each user@domain derived from this directory is available to all components in the system—it's a single centralized directory for storing all of your user information. If you register as a user with a SIP phone or if you try to leave a voicemail message for a user, FreeSWITCH looks in the same place for user data. This is important because it limits duplication of data, and makes it more efficient than it would be if each component kept track of its users separately.

This system should work well for a small system with a few users in it, but what about a large system with thousands of users? What if a user wants to connect his existing database to FreeSWITCH to provide the user directory? Well, using mod_xml_curl (download here-ch:1,ch:3), we can create a web service that gets the request for the entries in the user directory, in the same way a web page sends the results of a form submission. In turn, that web service can query an existing database of users formatted any way possible, and construct the XML records in the format that FreeSWITCH registry expects. mod_xml_curl returns the data to the module requesting the lookup. This means that instant, seamless integration with your existing setup is possible; your data is still kept in its original, central location.

The user directory can be accessed by any subsystem within FreeSWITCH. This includes modules, scripts, and the FSAPI interface among others. In this article, we are going to learn how the Sofia SIP module employs the user directory to authenticate your softphone or hardware SIP phone. If you are a developer you may appreciate some nifty things you can do with your user directory, such as adding a <variables> element to either the <domain>, the <groups>, or the <user> element. In this element you can set many <variable> elements, allowing you to set channel variables that will apply to every call made by a particular authenticated user. This can come in very handy in the Dialplan because it allows you to make user-specific routing decisions. It is also possible to define IP address ranges using CIDR notation, which can be used to authenticate particular users based on what remote network address they connect from. This removes the need for a login and password, if your user always logs in from the same remote IP address.

The directory is implemented in pure XML. This is advantageous for several reasons, not the least of which is the "X" in XML: Extensible. Since XML is, by definition, extensible, the directory structure is also extensible. If we need to add a new element into the directory, we can do so simply by adding to the existing XML structure.

Authentication versus authorization
Authentication is the process of identifying a user. Authorization is the process of determining the level of access of a user. Authentication answers the question, "Is this person really who he says he is?" Authorization answers the question, "What is this person allowed to do here?" When you see expressions such as "IP Auth" and "Digest Auth", remember that they are referring to the two primary ways of identifying (that is, authenticating) a user. IP authorization is based upon the user's IP address. Digest authentication is based upon the user supplying a username and password. SIP (and FreeSWITCH) can use either method. Visit http://en.wikipedia.org/wiki/Digest_access_authentication for a discussion of how digest authentication works

Working with the FreeSWITCH user directory

The default configuration has one domain with a directory of 20 users. Users can be added or removed very easily. There is no set limit to how many users can be defined on the system. The list of users is collectively referred to as the directory. Users can belong to one or more groups. Finally, all the users belong to a single domain. By default, the domain is the IP address of the FreeSWITCH server.

In the following sections we will discuss these topics:

  • User features
  • Adding a user
  • Testing voicemail
  • Groups of users

User features

Let's begin by looking at the XML file that defines a user. Locate the file conf/directory/default/1000.xml and open it in an editor. You should see a file like the following:

<include>
<user id="1000">
<params>
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow"
value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension
1000"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>

The XML structure of a user is simple. Within the <include> tags the user has the following:

  • The user element with the id attribute
  • The params element, wherein parameters are specified
  • The variables element, wherein channel variables are defined

Even before we know what much of the specifics mean, we can glean from this file that the user id is 1000 and that there is both a password and a vm-password. In this case, the password parameter refers to the SIP authorization password. The expression $${default_password} refers to the value contained in the global variable default_password which is defined in the conf/vars.xml file. If you surmised that vm-password means "voicemail password" then you are correct. This value refers to the digits that the user needs to dial when logging in to check his or her voicemail messages. The value of id is used both as the authorization username and the SIP username.

Additionally, there are a number of channel variables that are defined for this user. Most of these are directly related to the default Dialplan. The following table lists each variable and what it is used for:

Variable

Purpose

toll_allow

Specifies which types of calls this user can make

accountcode

Arbitrary value that shows up in CDR data

user_context

The Dialplan context that is used when this person makes a phone call

effective_caller_id_name

Caller ID name displayed on called party's phone when calling another registered user

effective_caller_id_number

Caller ID number displayed on called party's phone when calling another registered user

outbound_caller_id_name

Caller ID name sent to provider on outbound calls

outbound_caller_id_number

Caller ID number sent to provider on outbound calls

callgroup

Arbitrary value that can be used in Dialplan or CDR

In summary, a user in the default configuration has the following:

  • A username for SIP and for authorization
  • A voicemail password
  • A means of allowing/restricting dialling
  • A means of handling caller ID being sent out
  • Several arbitrary variables that can be used or ignored as needed

Let's now add a new user to our directory.

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(For more resources on Telephony, see here.)

Adding a user

Adding one or more users is a simple two-step process, which is as follows:

  • Create a new XML file for the user, usually by copying an existing file
  • Modify the Local_Extension Dialplan entry

In this example, we will create a new user for a person named "Gwen" and a username of "1100". Follow these steps:

  • Open a terminal window, change directory to conf/directory/default
  • Make a copy of 1000.xml and name it 1100.xml. A Linux/Unix session looks like the following:

    #>cd /usr/local/freeswitch/conf/directory/default
    #>cp 1000.xml 1100.xml

  • Open 1100.xml in an editor and make the following changes:
    • Replace all occurrences of "1000" with "1100"
    • Change the value of effective_caller_id_name to "Gwen"

The new file should look like the following:

<include>
<user id="1100">
<params>
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="1100"/>
</params>
<variables>
<variable name="toll_allow"
value="domestic,international,local"/>
<variable name="accountcode" value="1100"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Gwen"/>
<variable name="effective_caller_id_number" value="1100"/>
<variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>

Save the file. Next, we need to edit the Dialplan entry for Local_Extension. Open conf/dialplan/default.xml in an editor and locate the following lines:

<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">

This Dialplan extension, as its name implies, routes calls to local extensions. In our case, a local extension is a phone registered to a user in our directory. Recall that FreeSWITCH comes with 20 directory users predefined, numbered 1000 through 1019. This extension corresponds to those 20 users. By default, any call made to 1000, 1001, … 1019 will be handled by the Local_Extension Dialplan entry. We need to add "1100" to the regular expression. Edit the expression value so that it looks like the following:

^(10[01][0-9]|1100)$

Save the file.

The last thing we need to do is reload the XML configuration. Launch fs_cli and issue the reloadxml command as follows:

freeswitch@internal> reloadxml
+OK [Success]

freeswitch@internal> 2009-11-20 16:47:36.986620 [INFO] mod_enum.c:808 ENUM Reloaded
2009-11-20 16:47:36.986620 [INFO] switch_time.c:661 Timezone reloaded 530 definitions

Linux/Unix users can save time by opening two terminal windows. Run fs_cli in one window and your editor in the other. For advanced handling of multiple windows check out the GNU Screen utility. For more information check: http://www.gnu.org/software/screen.

Our new extension has been defined, and we should now be able to register a SIP phone to user 1100. Register a SIP phone to user 1100(Describing the methods for registration is out of the scope of this article). An X-Lite configuration looks like this:

The registered phone can now make outbound calls, and can receive inbound calls from those who dial 1100.

To see which SIP phones are registered issue this command at the FreeSWITCH command line: sofia status profile internal.

Now that we have successfully added a user, let's test a common feature: voicemail.

Testing voicemail

Each user in the directory has a voice "mailbox" where others can leave voice messages. By default, unanswered calls to a user will go to the user's voicemail after 30 seconds. Make a test call to confirm that everything is working. Dial the destination extension and let it ring. After about 30 seconds the voicemail system will answer; record a message of at least three seconds (the minimum message length), and then hang up. (If you have only one phone for testing, then try dialing your own extension.) The user's phone will now have a message-waiting indicator. An X-Lite softphone with a message waiting looks like the following image:

Notice the envelope icon and the red telephone icon. These indicate a new message waiting and a missed call, respectively.

Save time when leaving a voice message by pressing #, to skip past the user's outbound greeting.

Retrieving the message is also simple: Dial *98 or 4000. The voicemail system will guide you through logging in and listening to new or saved messages. A typical session would look like the following:

4000

"Welcome to your voicemail. Please enter your ID, followed by the
pound sign."

1100#

"Please enter your password, followed by the pound sign."

1100#

"You have one new message."

When a user has a new message, the system will automatically play it along with the date and time that the message was left. The default voicemail menus are configured as follows:

Main Menu

1—Listen to new messages
2—Listen to saved messages
5—Options menu (recorded name, greeting, and so on)
#—Exit voicemail

While Listening to a Message

1—Replay message from the beginning
2—Save message
4—Rewind
6—Fast-forward
0—Pause playback

After Listening to a Message

1—Replay message from the beginning
2—Save message
4—Send to e-mail (requires configuration)
7—Delete message
8—Undelete message

Feel free to try out some of these options. Log in to your voicemail and record an outbound greeting. By default you can record up to ten different greetings; however, most users only record a single greeting. Typically we will use greeting number one.

All of the voicemail options are customizable. Look in the file conf/autoload_configs/voicemail.conf.xml. You can edit the default voicemail profile or even create your own custom voicemail profiles.

Now that we have voicemail working, we can concentrate on one other useful feature: groups of users.

Groups of users

Larger installations frequently need the ability to dial multiple telephones. For example, a department in a company might have several users, all of whom are responsible for answering calls to that department. At the same time, they each have their own extension number, so they may individually receive calls. FreeSWITCH has a directory feature that allows users to be grouped together. A user can belong to multiple groups.

Some PBX systems employ an advanced form of inbound call routing called ACD or Automatic Call Distribution. Call groups are not used for this kind of application. Although it is beyond the scope of this publication, FreeSWITCH users wanting advanced functionality are encouraged to investigate FIFO queues. See http://wiki.freeswitch.org/wiki/Mod_fifo for more information.

Groups are defined in the file conf/directory/default.xml. Open the file and locate the groups node. Notice that there are four groups already defined. They are as follows:

  • Default—All users in the directory
  • Sales—1000 to 1004
  • Billing—1005 to 1009
  • Support—1010 to 1014

The latter three groups are merely arbitrarily defined groups that can be modified or removed as needed. The default group, though, is a bit more interesting. It contains every user in the directory. (Use with caution!) Let's add a new group and then examine how groups work:

  1. Add the following lines inside the groups node:

    <group name="custom">
    <users>
    <user id="1000" type="pointer"/>
    <user id="1100" type="pointer"/>
    </users>
    </group>

  2. If you have two or more telephones registered then use their extension numbers instead of 1000 and 1100. Save the file.
  3. Launch fs_cli and issue the reloadxml command

Confirm that the new custom group has been added by using the group_call command. Your output should be similar to the following:

freeswitch@internal> group_call custom

[presence_id=1000@10.15.0.91]error/user_not_registered,[presence_
id=1100@10.15.0.91]sofia/internal/sip:1100@10.15.0.136:31354;rinstance=
5e39ede358ffc0c8

What significance does this chunk of apparently random gibberish hold? The group_call command is used to create a dialstring for calling multiple telephones. In our example, user 1000 is not registered and therefore would not receive a call. (Hence the "error" of user_not_registered.) However, user 1100 is indeed registered. If a user in a group is not registered, then when the group is called, that user is effectively ignored. Before we can call our new group we need to add it to the Dialplan, as follows:

  1. Open conf/dialplan/default.xml and locate the group_dial_billing extension shown as follows:

    <extension name="group_dial_billing">
    <condition field="destination_number" expression="^2002$">
    <action application="bridge"
    data="group/billing@${domain_name}"/>
    </condition>
    </extension>

  2. Insert the following new lines after the </extension> tag of the group_dial_billing extension:

    <extension name="group_dial_custom">
    <condition field="destination_number" expression="^2003$">
    <action application="bridge"
    data="group/custom@${domain_name}"/>
    </condition>
    </extension>

  3. Save the file.
  4. Launch fs_cli and issue the reloadxml command.
  5. Test your group by dialing 2003. All the extensions in your group should ring.

When all of the phones in a group are ringing, the first one to answer will "win" and receive the call. All the other phones will stop ringing.

We have seen how we can connect telephones to FreeSWITCH, as well as the many features they have. Now let's discuss how to make phone calls outside the local FreeSWITCH server.

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(For more resources on Telephony, see here.)

Connecting to the world with gateways

The counterpart to having a user register to your FreeSWITCH server is to have your server register as a user on a remote server. This is accomplished using gateways. A gateway is quite simply an outbound registration to another SIP server. Telephone service providers use very large servers to provide SIP trunks to their subscribers. In FreeSWITCH, we can use a gateway to connect to a SIP provider. We can also use a gateway to connect to another SIP server, such as another FreeSWITCH server or any SIP-compliant IP-PBX.

Setting up a new gateway

A gateway simply connects to a SIP server just like a SIP phone connects to FreeSWITCH. As such, a gateway configuration bears some resemblance to a SIP phone configuration. Like a SIP phone registering to FreeSWITCH, a gateway has some minimum requirements. They are as follows:

  • Username and password
  • Server address or IP, and port

These values are supplied by the service provider. Occasionally there are other parameters, like a proxy server and port. If you already have an account with a SIP provider then you can use it for your gateway. In this example, we will use an account from iptel.org.

Visit http://www.iptel.org to sign up for a free SIP account.

To add a new gateway, follow these steps:

  1. Create a new XML file in conf/sip_profiles/external. This example will use iptel.org.xml. Add the following lines, inserting the proper values for your provider:

    <include>
    <gateway name="iptel">
    <param name="username" value="MY_USER_NAME"/>
    <param name="password" value="MY_PASSWORD"/>
    <param name="realm" value="iptel.org"/>
    <!-- iptel.org requires a 'proxy' parameter -->
    <param name="proxy" value="sip.iptel.org"/>
    </gateway>-->
    </include>

  2. Save the file and then launch fs_cli.
  3. Issue the command /log 6 to decrease the verbosity of debug messages.
  4. Simply reloading the XML configuration will not add the new gateway. Issue the following command: sofia profile external restart reloadxml.
    The output will look as follows:

    Reload XML [Success]

    restarting: external
    freeswitch@internal> 2009-11-21 16:29:23.509986 [INFO] mod_enum.
    c:808 ENUM Reloaded

    2009-11-21 16:29:23.511578 [INFO] switch_time.c:661 Timezone
    reloaded 530 definitions

    2009-11-21 16:29:24.118566 [NOTICE] sofia_reg.c:85 UN-Registering
    iptel

    2009-11-21 16:29:24.713768 [NOTICE] sofia.c:1218 Waiting for
    worker thread

    2009-11-21 16:29:24.713768 [NOTICE] sofia_glue.c:3690 deleted
    gateway example.com

    2009-11-21 16:29:24.713768 [NOTICE] sofia_glue.c:3690 deleted
    gateway iptel


    2009-11-21 16:29:24.713768 [NOTICE] sofia_reg.c:2237 Added gateway
    'iptel' to profile 'external'

    2009-11-21 16:29:24.713768 [NOTICE] sofia_reg.c:2237 Added gateway
    'example.com' to profile 'external'

    2009-11-21 16:29:24.713768 [NOTICE] sofia.c:3149 Started Profile
    external [sofia_reg_external]

    2009-11-21 16:29:25.736445 [NOTICE] sofia_reg.c:333 Registering
    iptel

  5. Confirm that the gateway is registered properly. Issue the command sofia status. The output should look similar to the following:

The gateway's state should be REGED. If it says something else, like FAIL_WAIT, then most likely there is a configuration problem. Confirm your settings and try again.

Restarting a profile will disconnect all active calls that are currently routed through that profile. An alternate command to add a newly created gateway without restarting the entire profile is: sofia profile external rescan reloadxml

Now that our gateway is added, we need to modify the Dialplan so that we can make and receive calls.

Making calls

We will make a simple Dialplan entry that sends calls out our new gateway. Our new extension will accept the digit 9 and then the digit 1, followed by exactly ten more digits representing the telephone number to be dialed. (In a production environment there are many other possible strings, which can even be alphanumeric.)

To get started with making outbound calls, add your new extension to the Dialplan, following these steps:

  1. Create a new file in conf/dialplan/default named 01_custom.xml.
  2. Add the following text to the file:

    <include>
    <extension name="Dial Out Custom Gateway">
    <condition field="destination_number"
    expression="^9(1\d{10})$">
    <action application="bridge"
    data="sofia/gateway/custom/$1"/>
    </condition>
    </extension>
    </include>

  3. Save the file. Launch fs_cli and issue the reloadxml command.

The new extension is now ready to be tested. From a phone that is registered to FreeSWITCH, dial 9 plus a ten-digit phone number. For example, dial 9, 1-800-555-1212. It may take a moment for the call to be established. Confirm that audio is flowing in both directions and that each party can hear the other. If audio is flowing in only one direction, then most likely there is a problem with the NAT device on your local network.

Receiving calls

Generally, when you register your gateway with a SIP provider, the provider allows you to receive calls. (Telephones that register with FreeSWITCH are an example of this.) By default, FreeSWITCH treats incoming calls as inherently untrusted, even if they come from the opposite end of a registered gateway. These calls come into the "public" Dialplan context. From there they can be discarded or routed as needed. Let's set up a simple Dialplan entry to handling inbound calls to our iptel.org account.

  1. Create a new file in conf/dialplan/public named 01_iptel.xml.
  2. Add these lines to the file, using your account name as follows:

    <include>
    <extension name="iptel-inbound">
    <condition field="destination_number"
    expression="^(MY_IPTEL_USERNAME)$">
    <action application="set" data="domain_name==$${domain}"/>
    <action application="bridge" data="1000 XML default"/>
    </condition>
    </extension>
    </include>

  3. Save the file. Launch fs_cli and execute the reloadxml command.

Inbound calls will now be routed to extension 1000. You can route calls to any valid extension, including all the extensions tested.(The other tested extensions are out of the scope of this article).

Making calls without a gateway

Sometimes it is not necessary to use a gateway. For example, not all services require digest authorization. An example of this is the FreeSWITCH public conference server. In fact, the default Dialplan contains an extension for dialling the conference: 9888. (Actually, there are several different conference "rooms" on the public FreeSWITCH conference server.) Let's look at this extension. Open conf/dialplan/default.xml in an editor and locate the freeswitch_public_conf_via_sip extension. Note the bridge line:

<action application="bridge" data="sofia/${use_profile}/$1@conference.
freeswitch.org"/>

The value in ${use_profile defaults to "internal" (as defined in conf/vars.xml). When a user dials 9888 the dialstring that is sent out is actually as follows:

sofia/internal/888@conference.freeswitch.org

Notice that there is no mention of a gateway. Instead, FreeSWITCH simply sends the call out to the internal SIP profile. In other words, the local FreeSWITCH server sends a call to conference.freeswitch.org without actually authorizing it. This is possible because the server at conference.freeswitch.org does not require authorization for incoming calls. (This is where the gateway comes in—if the target server issues a challenge then the gateway will respond to that challenge with authorization credentials, namely the username and password.)

Not all SIP providers explicitly require digest authorization of calls; some perform IP authorization instead. In those cases you do not need to create a gateway. Instead, simply send the call out a SIP profile. Usually the internal SIP profile is sufficient for these kinds of calls.

SIP profiles and user a gents

Before we finish our discussion of SIP and the user directory, it would be good to touch upon a subject that some users initially find a bit daunting: SIP profiles. In the strictest sense of the word, a SIP profile in FreeSWITCH is a User Agent. In practical terms, this means that each SIP profile "listens" on a particular IP address and port number. The internal profile listens on port 5060, and the external profile listens on port 5080. Not only does the profile listen but it can respond as well. For example, when a phone sends a SIP REGISTER packet to FreeSWITCH (at port 5060), the internal profile "hears" the registration request and acts accordingly. The files in conf/sip_profiles/ are ones which determine how the profiles behave. Many of the parameters in these profiles are to customize how FreeSWITCH handles various SIP traffic scenarios. In most cases the defaults are reasonable and should work. In other cases, though, you may find that because of the peculiarities in various VoIP phones and vendors, you will need to make adjustments.

Lastly, do not let the profile names internal and external be a source of confusion. Each profile is simply a user agent that is streamlined for a specific purpose. The internal profile is optimized to handle telephone registrations and calls between registered phones. The external profile is optimized for outbound gateway connections and several NAT traversal scenarios.

For a deeper discussion of user agents and the concept of a back-to-back user agent (B2BUA) see http://en.wikipedia.org/wiki/Back-to-back_user_agent

Summary

In this article, we discussed the following:

  • How FreeSWITCH collects users into a directory
  • How FreeSWITCH uses a VoIP protocol, SIP, to connect users to each other, and to the world
  • SIP is similar to e-mail in that it has users and domains
  • Employing various user features like voicemail
  • Adding a new user and modifying the Dialplan accordingly
  • Connecting to the outside world with gateways
  • SIP profiles and user agents

In this article, we made some minor modifications to the default XML Dialplan, and we learned how to set up users and domains within the XML user directory. Now that we have a general understanding of how these modifications work, we can continue to build upon this foundation. We can now begin to form a much more detailed understanding of FreeSWITCH as we further explore the XML Dialplan module, the default and most commonly used call routing engine available in FreeSWITCH.


Further resources on this subject:


About the Author :


Anthony Minessale

Anthony Minessale has been working with computers for nearly 30 years. He is the primary author of FreeSWITCH and Director of Engineering for CudaTEL at Barracuda Networks.

He created and continues to run the ClueCon Telephony Developers Conference, held every August in Chicago.

He has extensive experience in the Internet industry and VoIP. Before creating FreeSWITCH, he contributed heavily to the Asterisk open source project, producing many features that are still in use today. At Barracuda Networks, Anthony oversees the production and development of the CudaTEL PBX appliance that uses FreeSWITCH as its core telephony engine.

Darren Schreiber

Darren Schreiber is the CEO and Co-founder of 2600 Hz. He began working heavily in open source voice with the FreeSWITCH project, where he engaged with Brian, Mike, and Anthony. His projects have since evolved into two enterprise VoIP platforms that allow a multitude of development of voice, SMS, and video applications to be delivered to customers.

He has 15 years of voice and IT experience including developing multiple enterprise SaaS infrastructures for hosting and remotely managing IT, voice, and e-commerce services. He is a guest lecturer at major universities on VoIP technology and leads paid international VoIP trainings. As a serious telephony enthusiast since a young age, he has worked extensively with VoIP technologies. He graduated from Rensselaer Polytechnic Institute with a degree in Computer Science and Business Management.

He is also a co-author of FreeSWITCH Cookbook, Packt Publishing.

Michael S. Collins

Michael Collins is a 15-year veteran of telecommunications and PBX systems. A relative newcomer to VoIP, he discovered FreeSWITCH in its very early stages and has been an enthusiastic community member ever since. He works with the core FreeSWITCH development team, assisting with documentation and community development. Michael currently works for a technology company that makes extensive use of FreeSWITCH.

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Active Directory Disaster Recovery


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