FreeSWITCH Cookbook: Written by members of the FreeSWITCH team, this is the ultimate guide to getting the most out of the platform. Stuffed with over 40 recipes, just about every angle is covered, from call routing to enabling text-to-speech conversion.

FreeSWITCH uses Perl-compatible regular expressions (PCRE) for pattern matching. Consider this dialplan excerpt:

<extension name="example">
  <condition field="destination_number" expression="^(10\d\d)$">
    <action application="log" data="INFO dialed number is [$1]"/>

This example demonstrates the most common uses of regular expressions in the dialplan: matching against the destination_number field (that is, the digits that the user dialed) and capturing the matched value in a special variable named $1. Let's say that a user dials 1025; our example extension would match 1025 against the pattern ^(10\d\d)$ and determine that this is indeed a match. All actions inside the condition tag would be executed. The action in our example would execute the log application. The log application will then print a message to the console, using the INFO log level, which, by default, will be in green text. The value in $1 is expanded (or interpolated) when printed out:

2011-01-09 13:38:31.864281 [INFO] mod_dptools.c:1152 dialed number is [1025]

Understanding these basic principles will enable you to create effective dialplan extensions. For more tips on using regular expressions, be sure to visit http://wiki.freeswitch.org/wiki/Regex.

As previously mentioned, contexts are logical groups of extensions. The default FreeSWITCH configuration contains three contexts:

Each of these contexts serves a purpose, and knowing about them will help you leverage their value for your needs.

<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>

<extension name="dx">
  <condition field="destination_number" expression="^dx$">
  ...

If you are using the default configuration, then users 1000 through 1019 are pre-configured, both in the directory and the dialplan. To create a user outside this range, it is generally easiest to just run the add_user script, found in the FreeSWITCH source directory under scripts/perl. For example, to add the user 1020, launch this script from the FreeSWITCH source directory, specifying the user ID on the command line:

scripts/perl/add_user 1020

You can also specify a range of users:

scripts/perl/add_user –-users=1020-1029

You will see a note about how many users were added. If you have the CPAN module Regexp::Assembly installed, then the script will also generate a 'sample regular expression pattern'. For our example, we will add a range of users 1020-1029.

Follow these steps:

The Local_Extension is the default dialplan entry that allows directory users to be called. Remember, simply adding a user to the directory does not mean that the user can be dialed. (It does, though, usually mean that the user can make outbound calls.) So in order for your new user to be reachable, you need to add his or her user ID to the dialplan. By default, Local_Extension has a regular expression that will match 1000, 1001, … 1019. When adding users outside that number range, it is necessary to modify the regular expression to account for those new numbers. In our example, we added user IDs 1020 through 1029, so we need to match those. We use this regular expression:

^(10[012][0-9])$

This matches 1000 through 1029. Let's say we added another block of user IDs with the range of 1030 through 1039. We could modify our regular expression to catch those as well:

^(10[0123][0-9])$

It is considered a best practice not to add a large range of dialable numbers in the Local_Extension without having the corresponding users in the directory. Doing so can make troubleshooting dialplan issues more difficult.

As a reminder, be sure to execute the reloadxml command each time you modify the regular expression (the changes you make to your XML configuration files will not take effect until they are loaded into memory, which is what reloadxml command does).

Routing a call requires two pieces of information—the phone number being routed and a destination for that phone number. In our example, we will use a DID number of 8005551212. Our destination will be user 1000. Replace these sample numbers with the appropriate values for your setup.

Follow these steps:

All calls that come in to the FreeSWITCH server from outside (as well as internal calls that are not authenticated) are initially handled in the public dialplan context (dialplan contexts were discussed in more detail in this chapter's introduction). Once the call hits the public context, we try to match the destination_number field. The destination_number is generally the DID number (see the There's more section below for some caveats). Once we match the incoming number, we then set the domain_name channel variable to the default domain value and then transfer the call to user 1000 (FreeSWITCH is domain-based in a way similar to e-mail.

Most systems have only a single domain, although FreeSWITCH supports multiple domains. See the FreeSWITCH wiki for explicit information on multiple domain configuration). The actual transfer happens with this dialplan entry:

<action application="transfer" data="1000 XML default"/>

In plain language, this tells FreeSWITCH to transfer the call to extension 1000 in the XML dialplan and the default context. The default context contains the Local_Extension, which handles calls to users' telephones.

Keep in mind the match in destination_number must match what the provider sends to FreeSWITCH, not necessarily what the calling party actually dialed. In North America, there are providers that send DID information in various formats such as:

The expression must match what the provider sends. One way to accomplish this is to have a few optional characters in the pattern. This pattern matches all three formats listed above:

<condition field="destination_number" expression="^\+?1?(8005551212)$">

The value \+? means optionally match a literal + character and the value 1? means "optionally match a literal digit 1". Now our pattern will match all three formats that are commonly used in North America (technically, our pattern will also match +8005551212, but we are not concerned about that. However, the pedantic admin might be, so he or she can use the pattern ^(\+1)?1?(8005551212)$ instead).

Making outbound calls requires you to know the numbering format that your provider requires. For example, do they require all 11 digits for US dialing? Or will they accept 10? In our example, we're going to assume that our provider will accept a 10-digit format for US dialing.

Routing outbound calls is simply a matter of creating a dialplan entry. Follow these steps:

Assuming you have a phone set up on the default context, our regular expression will match any destination_number that follows the US dialing format (10 or 11 digits) and send the call to our_sip_provider in a 10-digit format.

The regular expression matching in FreeSWITCH allows the possibility of having very powerful conditions. You can also match caller_id_number to route calls from a user at extension 1011 out to the second gateway called our_sip_provider2 and everyone else at the our_sip_provider. Consider the following alternative outbound_calls.xml file:

<include>
  <extension name="outbound_calls_from_1011">
    <condition field="caller_id_number" expression="^1011$"/>
    <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
      <action application="bridge" data="sofia/gateway/our_sip_provider2/$1"/>
    </condition>
  </extension>
  <extension name="outbound_calls">
    <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
      <action application="bridge" data="sofia/gateway/our_sip_provider/$1"/>
    </condition>
  </extension>
</include>

Note that we have two extensions. The first one tries to match the caller_id_number field to the value 1011. If it matches 1011, then the call gets sent out to the our_sip_provider2 gateway, otherwise the second extension is matched and the call goes out to the our_sip_provider gateway. Note that we use $1 to capture the matching value in the conditions' expressions. In each case, we capture exactly 10 digits which correspond to the area code (three digits), exchange (three digits), and phone number (four digits). These are North American Numbering Plan (NANPA) numbers. The regular expression used to capture dialed digits will vary depending upon the country.

Open conf/dialplan/default.xml in a text editor or create or edit a new XML file in the conf/dialplan/default/ subdirectory.

Add a comma-separated list of endpoints to your bridge (or originate) application. For example, to ring userA@local.pbx.com and userB@local.pbx.com simultaneously, use an extension like this:

<extension name="ring_simultaneously">
  <condition field="destination_number" expression="^(2000)$">
    <action application="bridge" data="{ignore_early_media=true}sofia/internal/userA@local.pbx.com,sofia/sip/userB@local.pbx.com"/> 
  </condition>
</extension>

Putting comma-separated endpoints in the argument to bridge causes all of the endpoints in that list to be dialed simultaneously. It sounds simple, however, there are several factors to consider when ringing multiple devices simultaneously in a real environment. The bridge application will connect the call to whoever sends media first. This includes early media (ringing). To put this another way, if you bridge a call to two parties and one party starts sending a ringing signal back to you, that may be considered media and the call will be connected to that party only. Ringing of the other phones will cease.

If you find that calls always go to a specific number on your list of endpoints versus ringing all numbers, or that all phones ring for a moment before ringing only a single number, your call may be getting bridged prematurely because of early media. Notice that we added ignore_early_media=true at the beginning of the dial string. As its name implies, ignore_early_media tells the bridge application not to connect the calling party to the called party when receiving early media (such as a ringing or busy signal). Instead, bridge will only connect the calling party to the called party who actually answers the call. In most cases, it is useful to ignore early media when ringing multiple endpoints simultaneously.

In some scenarios, you may also wish to ring specific devices for a limited amount of time. You can apply the leg_timeout parameter to each leg of the bridge to specify how long to ring each endpoint, like this:

<action application="bridge" data="[leg_timeout=20]sofia/internal/userA@local.pbx.com,[leg_timeout=30]sofia/sip/userB@local.pbx.com"/>

In this example, userA's phone would ring for a maximum of 20 seconds while userB's phone would ring for a maximum of 30 seconds.

This method of calling multiple parties works well for small numbers of endpoints. However, it does not scale to dozens or more users. Consider using a FIFO queue in such an environment (FreeSWITCH's mod_fifo is discussed at length online at http://wiki.freeswitch.org/wiki/Mod_fifo). See also Ringing multiple endpoints sequentially (simple failover) for an example of ringing a group of endpoints one at a time, which includes an expanded discussion of using call timeouts.

Open conf/dialplan/default.xml in a text editor or create or edit a new XML file in the conf/dialplan/default/ subdirectory.

Add a pipe-separated list of endpoints to your bridge (or originate) application. For example, to ring userA@local.pbx.com and userB@local.pbx.com sequentially, use an extension like this:

<extension name="ring_sequentially">
  <condition field="destination_number" expression="^(2001)$">
    <action application="bridge" data="{ignore_early_media=true}sofia/internal/userA@local.pbx.com|sofia/sip/userB@local.pbx.com"/> 
  </condition>
</extension>

Putting pipe-separated endpoints in the argument to bridge causes all of the endpoints in that list to be dialed sequentially. The first endpoint on the list that is successfully connected will be bridged and the other endpoints will not be dialed. There are several factors to consider when ringing multiple devices sequentially.

Notice that we added ignore_early_media=true at the beginning of the dial string. As its name implies, ignore_early_media tells the bridge application not to connect the calling party to the called party when receiving early media (such as a ringing or busy signal). Instead, bridge will only connect the calling party if the called party actually answers the call. In most cases you will need to ignore early media when dialing multiple endpoints sequentially.

Handling various failure conditions can be a challenge. FreeSWITCH has a number of options that lets you tailor bridge and originate to your specific requirements.

<action application="bridge" data="{ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}sofia/internal/userA@local.pbx.com|sofia/sip/userB@local.pbx.com"/>

condition_name:number_of_hits:tone_detect_frequencies

destination_out_of_order:2:1776.7. 

<action application="bridge" data="[leg_timeout=10]sofia/internal/userA@host|[leg_timeout=15]sofia/internal/userB@host"/>

<action application="bridge" data="{originate_continue_on_timeout=true}[leg_timeout=10]sofia/internal/userA@host|[leg_timeout=15]sofia/internal/userB@host"/>

<extension name="ring_sequentially">
  <condition field="destination_number" expression="^(2001)$">
    <action application="set" data="continue_on_fail=true"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data={ignore_early_media=true}sofia/internal/userA@local.pbx.com"/>
    <action application="log" data="INFO call to userA failed."/>
    <action application="bridge" data={ignore_early_media=true}sofia/internal/userB@local.pbx.com"/>
    <action application="log" data="INFO call to userB failed."/>
  </condition>
</extension>

The first thing you need to do to take advantage of enterprise originate is to fully understand the regular originate. Originate is the term used to indicate making an outbound call. Although there is an originate command that can be used at the fs_cli, the method by which you mostly use the originate command is with the bridge dialplan application.

You will need to open conf/dialplan/default.xml in a text editor or create or edit a new XML file in the conf/dialplan/default/ subdirectory.

The usage of enterprise originate is similar to the ring simultaneously example, but an alternate delimiter (:_:) is used:

<extension name="enterprise_originate">
  <condition field="destination_number" expression="^(2000)$">
    <action application="bridge" data="{ignore_early_media=true}sofia/internal/userA@local.pbx.com:_:{myoption=true}sofia/sip/userB@local.pbx.com"/> 
  </condition>
</extension>

<extension name="enterprise_originate2">
  <condition field="destination_number" expression="^(2001)$">
    <action application="bridge" data="{ignore_early_media=true}sofia/internal/userA@local.pbx.com,sofia/sip/userB@local.pbx.com:_:sofia/internal/userC@local.pbx.com,sofia/internal/userD@local.pbx.com"/> 
  </condition>
</extension>

The entire input string is broken up into smaller strings, based on the :_: symbol.

Each of those smaller strings is fed to the regular originate engine in parallel and the first channel to answer will be bridged to the caller. Once one endpoint answers, the rest of the calls in the enterprise will be canceled.

Enterprise originate has a few special aspects to consider when using it to place calls.

<action application="bridge" data="<ignore_early_media=true>{myvar=inner1}[who=userA]sofia/internal/userA@local.pbx.com,[who=userB]sofia/sip/userB@local.pbx.com:_:{myvar=inner2}[who=userC]sofia/internal/userC@local.pbx.com,[who=userD]sofia/internal/userD@local.pbx.com"/> 

To learn more about originate and enterprise originate, look at some of the other examples in this chapter and study the default dialplan distributed with FreeSWITCH. There are several examples of the many things you can do when placing outbound calls found in conf/dialplan/default.xml.

Determine the parameters for your routing. In this example, we will define business hours as Monday through Friday, 8AM to 5PM. Additionally, we will add a day_part variable to reflect morning (midnight to noon), afternoon (noon to 5PM), or evening (6PM to midnight).

Create an extension at the beginning of your dialplan by following these steps:

The Time of day , day of week setup extension defines two channel variables, namely, office_status and day_part. Note the use of inline="true" in our set applications. These allow for immediate use of the channel variables in later dialplan condition statements. Every call that hits this dialplan context will now have these two channel variables set (they will also show up in CDR records if you need them). You may have also noticed continue="true" in the extension tag and break="never" in the condition tags. These tell the dialplan parser to keep looking for more matches when it would otherwise stop doing so. For example, without continue="true" set, when the dialplan matched one of the conditions in the Time of day , day of week setup extension, then it would stop looking at any more extensions in the dialplan. In a similar way, the break="never" attribute tells the parser to keep looking for more conditions to match within the current extension (by default, when the parser hits a failed condition, it stops processing any more conditions within the current extension).

Our sample extension number is 5001. Note the action it takes:

<action application="execute_extension" data="5001_${office_status}"/>

This sends the call back through the dialplan looking for a destination_number of 5001_open or 5001_closed. We have defined both of those destinations with the extensions "office is open" and "office is closed," respectively. Now we can play different greetings to the caller—one for when the office is open and a different one for when the office is closed. As a nice touch, for all calls, we play a sound file that says, "Good morning", "Good afternoon", or "Good evening", depending on what value is in the channel variable day_part.

You may be wondering why we did not simply use a condition to test office_status for the value open and then use action tags for "office open" and anti-action tags for "office closed". There is nothing preventing us from doing this. However, what if you need to have an office status other than "open" or "closed"? For example, what if you have an office that needs to play a completely different greeting during lunch time? This is difficult to accomplish with only anti-action tags, but with our example, it is almost trivial. Let's make it a bit more challenging by adding a lunch period that runs from 11:30AM to 12:30PM. We cannot use hour="11.5-12.5", however, we do have another value we can test—time-of-day. This parameter lets us define periods in the day at a granularity of minutes or even seconds. The value range is 00:00 through 23:59 or 00:00:00 through 23:59:59. Consider this new Time of day, day of week setup snippet:

<extension name="Time of day, day of week setup" continue="true">
  <condition wday="2-6" hour="8-17" break="never">
    <action application="set" data="office_status=open" inline="true"/>
    <anti-action application="set" data="office_status=closed" inline="true"/>
  </condition>
  <condition wday="2-6" time-of-day="11:30-12:30" break="never">
    <action application="set" data="office_status=lunch" inline="true"/>
  </condition>

Notice that we need to explicitly define the weekend, since we cannot rely on a simple boolean open or closed condition. However, we now have a new office_status of lunch available to us. We define an additional extension to handle this case:

<extension name="office is at lunch">
  <condition field="destination_number" expression="^(5001_lunch)$">

Add the specific dialplan actions for handling calls during the office's lunch hour and you are done. You can add as many new office statuses as you need.

Refer to the XML dialplan wiki page (http://wiki.freeswitch.org/wiki/Dialplan_XML) for more information on the usage of break, continue, and inline attributes.

Be sure that you have your DIDs and users configured. In this example, we will use testuser as the username with a phone number of 4158867999 and our domain is my.phoneco.test.

Create a dialplan extension, specifically for handling calls to the DID number and use some regular expression syntax to parse out the information. Here is an example:

<extension name="call_4158867999">
  <condition field="destination_number" expression="^\+?1?4158867999$"/>
  <condition field="${sofia_contact(testuser@my.phoneco.test)}" expression="^[^\@]+(.*)">
    <action application="bridge" data="sofia/external/4158867999$1"/>
  </condition>
</extension>

You would typically make bridge calls to testuser using the bridge command with an argument of user/testuser. In this scenario, however, you wish to call testuser's registered endpoint but replace testuser with a phone number – 4158867999, in our example. To do this, you must retrieve testuser's current dialstring and remove the username, replacing it with the DID number.

In the example, we leverage the sofia_contact API and some regular expression magic. The first condition simply matches the user's DID phone number—we only want to act if the destination number is 4158867999. The interesting stuff happens in the second condition. The field is ${sofia_contact(testuser@my.phoneco.test)}. By wrapping an API call in ${}, the dialplan literally executes the API and uses the result as the field value. If we go to fs_cli and type sofia_contact testuser@my.phoneco.test, we get the result, which is something like this:

sofia/external/johndoe@12.34.56.7;fs_nat=yes

The regular expression pattern ^[^\@]+(.*) is applied against this value. The result is that everything after the @ is placed in the $1 variable. In our example, $1 contains @12.13.56.7;fs_nat=yes. Finally, we execute the bridge with the dialstring sofia/external/4158867999$1. With $1 expanded out, our destination is as follows:

sofia/external/4158867999@12.34.56.7;fs_nat=yes

We have successfully replaced testuser with 4158867999 while preserving the necessary IP address and parameters for contacting the server and sent the call to the proper destination.