Hello FreeSWITCHers! The FreeSWITCH team is working hard on getting version 1.8 ready for release. While you wait, we thought you might be interested in learning a few FreeSWITCH tricks you can try yourself. We had a look into our cookbook and found a few recipes you may be interested in. Have a look and don't forget you can learn even more about FreeSWITCH and other open source projects at ClueCon 2017 http://cluecon.com <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c372d&id=14ace00040&e=a45f2f7d64> Don't miss our early bird pricing available for only a short time! *Simple Call Failover* How often do you miss a customer’s phone call? It always seems like the phone rings at the most inconvenient time and ends just before you can get to it. Wouldn’t it be great if instead of going to voicemail it just rang your co-worker? Well, FreeSWITCH has a solution for you. What if you didn’t have to answer those calls by yourself? In the FreeSWITCH dialplan you can configure something called Simple Call Failover. What this means is a call comes in and rings multiple endpoints (two separate desk phones in this example, one right after the other) and the person that answers is connected to the call. That way if you are busy someone else can answer that call and keep the customers happy. To set this up open up your dialplan, in this example we will use the default located at conf/dialplan/default.xml in a text editor in our terminal. Navigate to the extension you would like to use and add these lines: <extension name="failover_example"> <condition field="destination_number" expression="^(1000)$"> <action application="bridge" data="{ignore_early_media= true}sofia/internal/first@domain.com|sofia/internal/second@domain.com"/> </condition> </extension> By adding a pipe between your endpoints you are telling FreeSWITCH to dial those endpoints sequentially. The first one that is successfully connected will get the call and FreeSWITCH will cease dialing the additional endpoints. Now, to do this FreeSWITCH needs to ignore any early media like ringing so we added ignore_early_media=true at the beginning to tell FreeSWITCH to only connect the call when it is answered. It is as easy as that. *Time of Day Routing* Did you know you can configure FreeSWITCH to route to certain extensions based on the time of day? This is particularly convenient if you want to play a different greeting based on your office’s hours. To configure this you first need to define some variables for FreeSWITCH to use as a reference. For this example, we will assume our office is open 8:00 a.m. to 4:00 p.m. We start by adding this extension to the beginning of your dialplan context: <extension name="Time of day, day of week example" continue="true"> <condition wday="2-6" hour="8-15" break="never"> <action application="set" data="office_status=open" inline="true"/> <anti-action application="set" data="office_status=closed" inline="true"/> </condition> </extension> Now you can use the previously defined variables to set your extensions. In the example below FreeSWITCH will first check to see if the office is open or closed based on the first condition we defined. After that, FreeSWITCH will then execute whichever dialplan extension is necessary depending on the value of ‘office_status’ in the first condition. This configuration allows you to set specific actions depending on the time of day and what you would like your IVR to do. In this example we are initiating a particular IVR depending on whether the office is open or closed. <extension name="Office_Hours, 2001_X"> <condition field="destination_number" expression="^(2001)$"> <action application="execute_extension" data="2001_${office_status}"/> </condition> </extension> <extension name="Open"> <condition field="destination_number" expression="^(2001_open)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="500"/><!-- Open IVR --> </condition> </extension> <extension name="Closed"> <condition field="destination_number" expression="^(2001_closed)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="500"/><!-- Closed IVR --> </condition> </extension> We have also added continue=”true” and break=”never” in our conditions. These tell FreeSWITCH to keep looking for more matches and without them FreeSWITCH would stop. Additionally, it should be noted that by setting variables we can easily allow for more conditions. If we were to set the “open” and “closed” extensions in an action tag and anti-action tag we are limited by only two options. In our current example you could go on to add things like lunch or holiday hours. *Manipulating SIP Request URIs* Another cool feature in FreeSWITCH is the ability to manipulate SIP To: Headers. You can use customized SIP To: Headers when routing calls to endpoints registered within your setup. This lets you send the DID in cases where a single registration is representing multiple DIDs. There isn’t a single flag to set in order to do this so you will have to configure your dialplan to grab the user’s registration information and parse out the username, replacing it with your custom setting. Be sure to only replace the username to prevent issues with things like NAT traversal. For this example, we have configured DID 123-456-7890, a username of user01, and domain of example.com. Our example extension will look like this: <extension name="call_1234567890"> <condition field="destination_number" expression="^(?:\+1|1)?1234567890$"/> <condition field="${sofia_contact(user01@example.com)}" expression="^[^\@]+(.*)"> <action application="bridge" data="sofia/external/123457890$1"/> </condition> </extension> This example shows the extension grabbing the user01 dial string and replacing the username with the preferred DID. In the first condition we are matching the determined DID and only if the incoming call matches that DID will the extension execute the actions. The second condition uses the results of an API, sofia_contact, to return the results as a value. This puts everything after the @ into the $1 variable which is then called during the bridge after our custom header. This preserves all the information required for contacting the server and sending the call to the correct destination. 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