Outgoing calls
In order to make your system useful, you need a way to dial out to the "real world". This recipe will cover dialing out to the PSTN and allow you to connect to landlines, cellular phones, and so on. In this recipe, we'll make an extension that will allow an outbound call to any valid US number. We'll attempt to complete the call using the gateway named our_sip_provider
(see the Configuring an SIP Gateway section in Chapter 2, Connecting Telephones and Service Providers).
Getting ready
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 (for example, without the international prefix "1").
How to do it...
Routing outbound calls is simply a matter of creating a dialplan entry:
- Create a new file in
conf/dialplan/default/
namedoutbound_calls.xml
. then add the following text:<include> <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>
- Save your XML file and issue the
reloadxml
command atfs_cli
.
How it works...
Assuming you have a phone set up on the default
context, your 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 format in regexp is as follows: optional "1", then one digit between 2 and 9, then two digits, then one digit between 2 and 9, and finally six digits. Only the part after the optional digit "1" is captured by the parentheses and passed down in the $1
variable.
There's more...
The regular expression matching in FreeSWITCH allows the privilege of having very powerful conditions. You can also match caller_id_number
to route calls from a user calling from extension 1011 out to the second gateway called our_second_sip_provider
, while everyone else will be sent through our_sip_provider
. Consider the following alternative outbound_calls.xml
file:
<include> <extension name="outbound_calls_from_1011"> <condition field="caller_id_number" expression="^1011$"/> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="bridge" data="sofia/gateway/our_second_sip_provider/$1"/> </condition> </extension> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <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 to the our_second_sip_provider
gateway. Otherwise, the next extension is matched and the call goes 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 (NANP) numbers. The regular expressions used to capture the format of dialed digits vary, depending upon the country.
Note
Regular expressions can be a challenge. There are a number of examples with explanations on the FreeSWITCH wiki. See http://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression for further details.
See also
- The Configuring an SIP phone to register with FreeSWITCH and Configuring an SIP gateway sections in Chapter 2, Connecting Telephones and Service Providers