Freeswitch Softphone
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Configuration
See mod_portaudio for initial setup information.
Originating Calls
This example API will dial the FreeSWITCH conference bridge from portaudio. One advantage of this method vs. using "pa call" is that a dialplan is not necessary.
originate {effective_caller_id_name=FreeSWITCH,effective_caller_id_number=9193869900}portaudio/auto_answer &bridge({originate_timeout=60}sofia/external/888@conference.freeswitch.org)
Receiving Calls
Create the following dialplan entry
<extension name="test">
<condition field="destination_number" expression="^3$">
<action application="bridge" data="portaudio"/>
</condition>
</extension>
And then call extension 3 with a softphone.
On the console you will see BRRRINNGGG and then type
pa answer
Music on Hold
Some config will need to be done to specify a default MOH file.
<document type="freeswitch/xml">
<!--
#set "pa_ring_file=C:\\freeswitch\\sounds\\_ring.wav"
-->
<!--
#set "pa_hold_file=C:\\freeswitch\\sounds\\_hold.wav"
-->
<section name="configuration" description="Various Configuration">
..
<variables>
<variable name="pa_ring_file" value="$${pa_ring_file}"/>
<variable name="pa_hold_file" value="$${pa_hold_file}"/>
FreeSWITCH as a Dedicated Soft Phone
The following example configuration assumes an extension of 1000, area code of 208, and that freeswitch.local domain is pointed to the internal IP of FreeSWITCH.
/conf/directory/default/freeswitch.local
Setup the gateway.
<include>
<user id="1000">
<gateways>
<gateway name="freeswitch.local">
<param name="username" value="1000"/>
<param name="password" value="1234"/>
<param name="from-user" value=""/>
<param name="from-domain" value="freeswitch.local"/>
<param name="expire-seconds" value="60"/>
<param name="register" value="true"/>
<param name="retry_seconds" value="30"/>
<param name="extension" value="1000"/>
<param name="context" value="default"/>
</gateway>
</gateways>
<params>
<param name="password" value="1234"/>
</params>
</user>
</include>
/conf/dialplan/default.xml
The example below assumes that the gateway is freeswitch.local which is the local phone system. In the example below make sure to change 1208 to your area code.
<?xml version="1.0" encoding="utf-8"?>
<include>
<context name="default">
<extension name="freeswitch.local">
<condition field="destination_number" expression="^(\d{4})$">
<action application="export" data="nolocal:absolute_codec_string=PCMU"/>
<action application="set" data="effective_caller_id_number=5555551212"/>
<action application="set" data="effective_caller_id_name=FreeSWITCH"/>
<action application="bridge" data="sofia/gateway/freeswitch.local/$1"/>
</condition>
</extension>
<extension name="freeswitch.local">
<condition field="destination_number" expression="^(\d{7})$">
<action application="export" data="nolocal:absolute_codec_string=PCMU"/>
<action application="set" data="effective_caller_id_number=5555551212"/>
<action application="set" data="effective_caller_id_name=FreeSWITCH"/>
<action application="bridge" data="sofia/gateway/freeswitch.local/1208$1"/>
</condition>
</extension>
<extension name="freeswitch.local">
<condition field="destination_number" expression="^(\d{10})$">
<action application="export" data="nolocal:absolute_codec_string=PCMU"/>
<action application="set" data="effective_caller_id_number=5555551212"/>
<action application="set" data="effective_caller_id_name=FreeSWITCH"/>
<action application="bridge" data="sofia/gateway/freeswitch.local/1$1"/>
</condition>
</extension>
<extension name="freeswitch.local">
<condition field="destination_number" expression="^(\d{11})$">
<action application="export" data="nolocal:absolute_codec_string=PCMU"/>
<action application="set" data="effective_caller_id_number=5555551212"/>
<action application="set" data="effective_caller_id_name=FreeSWITCH"/>
<action application="bridge" data="sofia/gateway/freeswitch.local/$1"/>
</condition>
</extension>
</context>
</include>
/conf/directory/default
Remove all local extensions .xml files or rename them with .noload
/conf/sip_profiles/outbound.xml
If FreeSWITCH is behind the same NAT then comment out ext-rtp-ip, and ext-sip-ip. "
<include>
<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="bridge" data="portaudio"/>
</condition>
</extension>
<extension name="public_did">
<condition field="destination_number" expression="^(1000)$">
<action application="bridge" data="portaudio"/>
</condition>
</extension>
</context>
</include>
