FreeSWITCH学习笔记：XML配置文件

本文更新于2023-11-17，使用FreeSWITCH 1.10.7。

加载顺序
autoload_configs/
dialplan/
- dialplan/default.xml
- dialplan/public.xml
directory/
- directory/default.xml
- directory/default/
  - directory/default/1000.xml
freeswitch.xml
ivr_menus/
- ivr_menus/demo_ivr.xml
lang
- lang/en
  - lang/en/demo
    - lang/en/demo/demo-ivr.xml
  - lang/en/en.xml
sip_profiles
vars.xml

FreeSWITCH默认的配置是一个SOHO PBX（家用电话小交换机）。

X-PRE-PROCESS标签是FreeSWITCH特有的，称为预处理指令，用于根据data参数设置（set）一些全局变量，以及将data参数指定的文件内容包含（include）到当前文件中。FreeSwitch在加载阶段只对其进行简单替换，并不进行语法分析，因此对它进行注释是没有效果的，解决办法是破坏X-PRE-PROCESS的定义（如将其替换为XPRE-PROCESS）。

可以$${VAR[:OFFSET[:LENGTH]]}形式引用全局变量，以${VAR[:OFFSET[:LENGTH]]}形式引用通道变量（局部变量）。OFFSET从0开始，如为负数表示从尾部开始。

加载顺序

配置文件的加载顺序如下：

 freeswitch.xml
  |- vars.xml
  |- autoload_configs/*.xml
  | [autoload_configs/acl.conf.xml]
  | [autoload_configs/callcenter.conf.xml]
  | [autoload_configs/cdr_csv_conf.xml]
  | [autoload_configs/cdr_pg_csv.conf.xml]
  | [autoload_configs/conference.conf.xml]
  | [autoload_configs/distributor.conf.xml]
  | [autoload_configs/event_socket.conf.xml]
  | [autoload_configs/fifo.conf.xml]
  | [autoload_configs/ivr.conf.xml]
  |    `- ivr_menus/*.xml
  |      [ivr_menus/demo_ivr.xml]
  | [autoload_configs/local_stream.conf.xml]
  | [autoload_configs/lua.conf.xml]
  | [autoload_configs/modules.conf.xml]
  | [autoload_configs/nibblebill.conf.xml]
  | [autoload_configs/post_load_modules.conf.xml]
  | [autoload_configs/sofia.conf.xml]
  |    `- sip_profiles/*.xml
  |      [sip_profiles/external.xml]
  |         `- sip_profiles/external/*.xml
  |           [sip_profiles/external/example.xml]
  |      [sip_profiles/internal.xml]
  | [autoload_configs/switch.conf.xml]
  | [autoload_configs/tts_commandline.conf.xml]
  | [autoload_configs/xml_cdr.conf.xml]
  | [autoload_configs/xml_curl.conf.xml]
  | [autoload_configs/xml_rpc.conf.xml]
  |- dialplan/*.xml
  | [dialplan/default.xml]
  |    `- dialplan/default/*.xml
  | [dialplan/public.xml]
  |    `- dialplan/public/*.xml
  |- directory/*.xml
  | [directory/default.xml]
  |    `- directory/default/*.xml
  |      [directory/default/1000.xml]
  `- lang/en/*.xml
    [lang/en/en.xml]
       `- lang/en/demo/*.xml
         [lang/en/demo/demo-ivr.xml]

重要的配置文件：

autoload_configs/modules.conf.xml
autoload_configs/sofia.conf.xml
autoload_configs/switch.conf.xml
dialplan/default.xml
dialplan/public.xml
directory/default.xml
directory/default/1000.xml
freeswitch.xml
sip_profiles/external.xml
sip_profiles/external/example.xml
sip_profiles/internal.xml
vars.xml

autoload_configs/

自动加载的模块的配置文件的目录。

一般来说每个模块有一个配置文件，文件名格式为“不包含mod_的模块名.conf.xml”，但文件名并不重要，可以改成其他的名字，只要扩展名为.xml。模块在启动时会向XML注册表中查找name为指定值的configuration，进而访问其下面的配置参数。

autoload_configs/acl.conf.xml

ACL权限控制配置。文件内容如下：

 <configuration name="acl.conf" description="Network Lists">
  <network-lists>
    <!--
         These ACL's are automatically created on startup.
 
         rfc1918.auto  - RFC1918 Space
         nat.auto      - RFC1918 Excluding your local lan.
         localnet.auto - ACL for your local lan.
         loopback.auto - ACL for your local lan.
    -->
 
    <list name="lan" default="allow">
      <node type="deny" cidr="192.168.42.0/24"/>
      <node type="allow" cidr="192.168.42.42/32"/>
    </list>
 
    <!--
        This will traverse the directory adding all users
        with the cidr= tag to this ACL, when this ACL matches
        the users variables and params apply as if they
        digest authenticated.
    -->
    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <!-- <node type="allow" cidr="192.168.0.0/24"/> -->
    </list>
 
  </network-lists>
</configuration>

list的name属性为名字，用于在其它地方引用。

autoload_configs/callcenter.conf.xml

呼叫中心模块配置。这是文件内容的一部分：

   <queues>
 
    <queue name="support@default">
      <param name="strategy" value="longest-idle-agent"/>
      <param name="moh-sound" value="$${hold_music}"/>
      <!--<param name="record-template" value="$${recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
      <param name="time-base-score" value="system"/>
      <param name="max-wait-time" value="0"/>
      <param name="max-wait-time-with-no-agent" value="0"/>
      <param name="max-wait-time-with-no-agent-time-reached" value="5"/>
      <param name="tier-rules-apply" value="false"/>
      <param name="tier-rule-wait-second" value="300"/>
      <param name="tier-rule-wait-multiply-level" value="true"/>
      <param name="tier-rule-no-agent-no-wait" value="false"/>
      <param name="discard-abandoned-after" value="60"/>
      <param name="abandoned-resume-allowed" value="false"/>
    </queue>
 
  </queues>
 
  <agents>
    <!--<agent name="1000@default" type="callback" contact="[leg_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />-->
  </agents>
  <tiers>
    <!-- If no level or position is provided, they will default to 1.  You should do this to keep db value on restart. -->
    <!-- <tier agent="1000@default" queue="support@default" level="1" position="1"/> -->
  </tiers>

queue为队列，name为队列的名字，用于在其它地方引用。

queue的param的name可为：

stragy：策略。其值可为：
- agent-with-fewest-calls：总时选择接电话次数最少的坐席。
- agent-with-least-talk-time：总是选择通话时间最短的坐席。
- long-idel-agent：选择空闲时间最长的坐席。
- random：随机选择。
- ring-all：所有坐席振铃，哪个先接就选哪个。
- round-robin：轮询。
- sequentially-by-agent-order：根据梯队和顺序选择。
- top-down：按固定的顺序选择。

agent为坐席，可使用以下属性：

busy-delay-time：如果坐席忙，再次选到该坐席的最短时长。
contact：坐席的呼叫字符串，用于在有来话时呼叫该坐席。
max-no-answer：坐席呼叫失败超过此次数，则将坐席状态设置为On Break。
name：坐席的名字，用于在其它地方引用。
reject-delay-time：坐席拒接后再次选到该坐席的最短时长。
status：坐席的初始状态。可为：
- Available：可用状态，可以接电话。
- Available (On Demand)：一种特殊的可用状态。
- Logged Out：退出服务状态。
- On Break：坐席已登录，但不可以接电话。
type：坐席的类型。可为：
- callback：onhook坐席。
- uuid-standby：offhook坐席。

tire为梯队，将队列与坐席关联起来。

autoload_configs/cdr_csv_conf.xml

使用CSV文件存储话单的配置。文件内容如下：

 <configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- 'cdr-csv' will always be appended to log-base -->
    <!--<param name="log-base" value="/var/log"/>-->
    <param name="default-template" value="example"/>
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
    <param name="rotate-on-hup" value="true"/>
    <!-- may be a b or ab -->
    <param name="legs" value="a"/>
        <!-- Only log in Master.csv -->
        <!-- <param name="master-file-only" value="true"/> -->
  </settings>
  <templates>
    <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
    <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
    <template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
    <template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
    <template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
    <template name="opencdrrate">"${uuid}","${signal_bond}","${direction}","${ani}","${destination_number}","${answer_stamp}","${end_stamp}","${billsec}","${accountcode}","${userfield}","${network_addr}","${regex('${original_caller_id_name}'|^.)}","${sip_gateway_name}"</template>
  </templates>
</configuration>

param的name可为：

default-template：默认模板。
legs：话单记录的腿（a腿/b腿）。
master-file-only：是否只记录话单汇总文件，不对每个分机单独记录话单文件。
rotate-on-hup：当FreeSWITCH收到SIGHUP信号时，是否将话单轮替。

autoload_configs/cdr_pg_csv.conf.xml

使用PostgreSQL存储话单的配置。文件内容如下：

 <configuration name="cdr_pg_csv.conf" description="CDR PG CSV Format">
  <settings>
    <!-- See parameters for PQconnectdb() at http://www.postgresql.org/docs/8.4/static/libpq-connect.html -->
    <param name="db-info" value="host=localhost dbname=cdr connect_timeout=10" />
    <!-- CDR table name -->
    <!--<param name="db-table" value="cdr"/>-->
 
    <!-- Log a-leg (a), b-leg (b) or both (ab) -->
    <param name="legs" value="a"/>
 
    <!-- Directory in which to spool failed SQL inserts -->
    <!-- <param name="spool-dir" value="$${log_dir}/cdr-pg-csv"/> -->
    <!-- Disk spool format if DB connection/insert fails - csv (default) or sql -->
    <param name="spool-format" value="csv"/>
    <param name="rotate-on-hup" value="true"/>
 
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
  </settings>
  <schema>
    <field var="local_ip_v4"/>
    <field var="caller_id_name"/>
    <field var="caller_id_number"/>
    <field var="destination_number"/>
    <field var="context"/>
    <field var="start_stamp"/>
    <field var="answer_stamp"/>
    <field var="end_stamp"/>
    <field var="duration" quote="false"/>
    <field var="billsec" quote="false"/>
    <field var="hangup_cause"/>
    <field var="uuid"/>
    <field var="bleg_uuid"/>
    <field var="accountcode"/>
    <field var="read_codec"/>
    <field var="write_codec"/>
    <!-- <field var="sip_hangup_disposition"/> -->
    <!-- <field var="ani"/> -->
  </schema>
</configuration>

param的name可为：

db-info：PostgreSQL标准的连接字符串。
db-table：数据库表名。

schema配置通道变量与数据库中字段的对应关系。

field的属性有：

column：数据库表的字段名。缺省则与var相同。
quote：SQL语句中的值是否使用单引号括起（如当值为整数时为false）。
var：通道变量。

autoload_configs/conference.conf.xml

多人电话会议配置。这是文件内容的一部分：

   <caller-controls>
    <group name="default">
      <control action="mute" digits="0"/>
      <control action="deaf mute" digits="*"/>
      <control action="energy up" digits="9"/>
      <control action="energy equ" digits="8"/>
      <control action="energy dn" digits="7"/>
      <control action="vol talk up" digits="3"/>
      <control action="vol talk zero" digits="2"/>
      <control action="vol talk dn" digits="1"/>
      <control action="vol listen up" digits="6"/>
      <control action="vol listen zero" digits="5"/>
      <control action="vol listen dn" digits="4"/>
      <control action="hangup" digits="#"/>
    </group>
  </caller-controls>
 
  <profiles>
    <profile name="default">
      <param name="rate" value="8000"/>
    </profile>
 
    <profile name="wideband">
    </profile>
 
    <profile name="ultrawideband">
    </profile>
 
    <profile name="cdquality">
    </profile>
  </profiles>

control的action属性可为：

execute_application：执行APP。control的data属性为执行APP的语句。

param的name可为：

alone-sound：会议只剩一个用户时向其播放的声音文件。
auto-record：录音模板。当会议中有至少两个人时就会自动开始录音。
comfort-noise：是否产生舒适噪声。
conference-flags：会议标志。rfc-4579为RFC4579定义的视频会议控制标准。
muted-sound：用户被禁言时向其播放的声音文件。
rate：采样率。
unmuted-sound：用户被取消禁言时向其播放的声音文件。

autoload_configs/distributor.conf.xml

号码连选模块配置。文件内容如下：

 <configuration name="distributor.conf" description="Distributor Configuration">
  <lists>
    <!-- every 10 calls to test you will get foo1 once and foo2 9 times...yes NINE TIMES! -->
    <!-- this is not the same as 100 with 10 and 90 that would do foo1 10 times in a row then foo2 90 times in a row -->
    <list name="test">
      <node name="foo1" weight="1"/>
      <node name="foo2" weight="9"/>
    </list>
  </lists>
</configuration>

autoload_configs/event_socket.conf.xml

Event Socket配置。文件内容如下：

 <configuration name="event_socket.conf" description="Socket Client">
  <settings>
    <param name="nat-map" value="false"/>
    <param name="listen-ip" value="::"/>
    <param name="listen-port" value="8021"/>
    <param name="password" value="ClueCon"/>
    <!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
    <!--<param name="stop-on-bind-error" value="true"/>-->
  </settings>
</configuration>

autoload_configs/fifo.conf.xml

呼叫队列模块配置。文件内容如下：

 <configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <param name="delete-all-outbound-member-on-startup" value="false"/>
  </settings>
  <fifos>
    <fifo name="cool_fifo@$${domain}" importance="0">
      <!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
    </fifo>
  </fifos>
</configuration>

param的name可为：

odbc-dsn：数据库ODBC的DSN。格式为：[odbc://]NAME:[USERNAME]:[PASSWORD]。

fifos用于配置静态坐席。

fifo的name属性为队列的名字，用于在其它地方引用。

每个member相当于一个坐席，其值为该坐席的呼叫字符串。坐席会轮流振铃（拒接其中一个，另一个就会振铃）。

timeout属性为呼叫超时的秒数。

simo属性为最大能服务的呼叫的数量。

lag属性为该坐席接听一个呼叫后再接听下一个呼叫的间隔秒数。

autoload_configs/ivr.conf.xml

IVR模块配置。文件内容如下：

 <configuration name="ivr.conf" description="IVR menus">
  <menus>
    <X-PRE-PROCESS cmd="include" data="../ivr_menus/*.xml"/>
  </menus>
</configuration>

autoload_configs/local_stream.conf.xml

本地文件流的配置。这是文件内容的一部分：

 <configuration name="local_stream.conf" description="stream files from local dir">
  <!-- fallback to default if requested moh class isn't found -->
  <directory name="default" path="$${sounds_dir}/music/8000">
    <param name="rate" value="8000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>
    <!-- list of short files to break in with every so often -->
    <!--<param name="chime-list" value="file1.wav,file2.wav"/>-->
    <!-- frequency of break-in (seconds)-->
    <!--<param name="chime-freq" value="30"/>-->
    <!-- limit to how many seconds the file will play -->
    <!--<param name="chime-max" value="500"/>-->
  </directory>
 
  <directory name="moh/8000" path="$${sounds_dir}/music/8000">
    <param name="rate" value="8000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>
  </directory>
</configuration>

name为名字，斜杠前面用于在其它地方引用，斜杠后面用于自动匹配采样率。如果找不到指定的流则会使用default流，如果default流也不存在则会提示错误。

path为声音文件的目录。

autoload_configs/lua.conf.xml

Lua语言配置。文件内容如下：

 <configuration name="lua.conf" description="LUA Configuration">
  <settings>
 
    <!-- 
    Specify local directories that will be searched for LUA modules
    These entries will be pre-pended to the LUA_CPATH environment variable
    -->
    <!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
    <!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->
 
    <!-- 
    Specify local directories that will be searched for LUA scripts
    These entries will be pre-pended to the LUA_PATH environment variable
    -->
    <!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
    <!-- <param name="script-directory" value="$${script_dir}/?.lua"/> -->
 
    <!--<param name="xml-handler-script" value="/dp.lua"/>-->
    <!--<param name="xml-handler-bindings" value="dialplan"/>-->
 
    <!--
        The following options identifies a lua script that is launched
        at startup and may live forever in the background.
        You can define multiple lines, one for each script you 
        need to run.
    -->
    <!--<param name="startup-script" value="startup_script_1.lua"/>-->
    <!--<param name="startup-script" value="startup_script_2.lua"/>-->
    
    <!--<hook event="CUSTOM" subclass="conference::maintenance" script="catch-event.lua"/>-->
  </settings>
</configuration>

param的name可为：

startup-script：FreeSWITCH启动时执行的脚本文件。
xml-handler-bindings：使用XML绑定的Section。Section见freeswitch.xml。
xml-handler-script：XML绑定执行的脚本文件。

autoload_configs/modules.conf.xml

配置FreeSWITCH启动时自动加载哪些模块。这是文件内容的一部分：

 <configuration name="modules.conf" description="Modules">
  <modules>
    <!-- Loggers (I'd load these first) -->
    <load module="mod_console"/>
    <!-- <load module="mod_graylog2"/> -->
    <load module="mod_logfile"/>
    <!-- <load module="mod_syslog"/> -->
  </modules>
</configuration>

autoload_configs/nibblebill.conf.xml

预付费计费模块配置。这是文件内容的一部分：

   <settings>
    <param name="odbc-dsn" value="bandwidth.com"/>
  </settings>

param的name可为：

db_column_account：计费账号对应的数据库表字段名。
global_heartbeat：通话过程中定时扣款的时间间隔。
odbc-dsn：数据库ODBC的DSN。格式为：[odbc://]NAME:[USERNAME]:[PASSWORD]。

autoload_configs/post_load_modules.conf.xml

其格式和用法与autoload_configs/modules.conf.xml差不多，不同的是其中定义的模块加载时间较晚。

autoload_configs/sofia.conf.xml

SIP模块配置。文件内容如下：

 <configuration name="sofia.conf" description="sofia Endpoint">
 
  <global_settings>
    <param name="log-level" value="0"/>
    <!-- <param name="abort-on-empty-external-ip" value="true"/> -->
    <!-- <param name="auto-restart" value="false"/> -->
    <param name="debug-presence" value="0"/>
    <!-- <param name="capture-server" value="udp:homer.domain.com:5060"/> -->
    
    <!-- 
        the new format for HEPv2/v3 and capture ID    
        
        protocol:host:port;hep=2;capture_id=200;
 
    -->
    
    <!-- <param name="capture-server" value="udp:homer.domain.com:5060;hep=3;capture_id=100"/> -->
  </global_settings>
 
  <!--
      The rabbit hole goes deep.  This includes all the
      profiles in the sip_profiles directory that is up
      one level from this directory.
  -->
  <profiles>
    <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
  </profiles>
 
</configuration>

各参数的含义如下：

capture-server：Homer方式抓包的Capture Node（又称Capture Server）的地址。

autoload_configs/switch.conf.xml

SIP核心配置。这是文件内容的一部分：

 <configuration name="switch.conf" description="Core Configuration">
 
  <settings>
    <param name="max-sessions" value="1000"/>
  </settings>
 
</configuration>

param的name可为：

core-db-dsn：core数据库的DSN。格式为：[odbc://]NAME:[USERNAME]:[PASSWORD]。
max-sessions：呼叫最大并发数。
rtp-end-port：RTP结束端口（含）。
rtp-start-port：RTP起始端口（含）。
sessions-per-second：每秒最大呼叫数。

autoload_configs/tts_commandline.conf.xml

调用其他TTS软件命令行的配置。文件内容如下：

 <configuration name="tts_commandline.conf" description="TextToSpeech Commandline configuration">
    <settings>
        <!--
        Some variables will be replaced :
        ${text}: input text (quoted)
        ${rate}: sample rate (example: 8000)
        ${voice}: voice_name passed to TTS(quoted)
        ${file}: output file (quoted, including .wav extension)
    
    Example commands can be found at:
    https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Examplecommands
        -->
        <param name="command" value="echo ${text} | text2wave -f ${rate} > ${file}"/>
    </settings>
</configuration>

在调用该模块时会自动设置以下变量：

file：声音文件名。默认扩展名为.wav，使用完毕会自动删除.wav文件。
rate：Channel的采样率。
text：TTS文本。
voicd：嗓音的名字。

autoload_configs/xml_cdr.conf.xml

生成XML格式的话单发送至外部HTTP服务器的配置。这是文件内容的一部分：

   <settings>
    <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
  </settings>

param的name可为：

url：话单的HTTP POST请求的URL地址。

autoload_configs/xml_curl.conf.xml

请求外部HTTP服务器生成XML的配置。这是文件内容的一部分：

 <configuration name="xml_curl.conf" description="cURL XML Gateway">
  <bindings>
    <binding name="example">
      <!-- <param name="gateway-url" value="http://www.freeswitch.org/gateway.xml" bindings="dialplan"/> -->
    </binding>
  </bindings>
</configuration>

param的name可为：

gateway-url：HTTP网关服务器的URL。bindings表示应用的Section。

autoload_configs/xml_rpc.conf.xml

FreeSWITCH Portal配置。文件内容如下：

 <configuration name="xml_rpc.conf" description="XML RPC">
  <settings>
    <!-- The port where you want to run the http service (default 8080) -->
    <param name="http-port" value="8080"/>
    <!-- if all 3 of the following params exist all http traffic will require auth -->
    <param name="auth-realm" value="freeswitch"/>
    <param name="auth-user" value="freeswitch"/>
    <param name="auth-pass" value="works"/>
 
 
    <!-- regex pattern to match against commands called against this service.
         If a command with arguments matches, it will be logged at INFO level -->
    <!--<param name="commands-to-log" value=""/> -->
 
  </settings>
</configuration>

dialplan/

拨号计划的配置目录。支持多个Context，每个Context可以写到一个XML文件中。

由于在处理Dialplan时要对每一项进行正则表达式匹配，这是非常影响效率的，所以在生产环境中往往要删除默认的Dialplan。

dialplan/default.xml

默认Context，一般用于内部注册用户路由。这是文件内容的一部分：

   <context name="default">
 
    <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="global" continue="true">
      <condition field="${call_debug}" expression="^true$" break="never">
        <action application="info"/>
      </condition>
 
      <condition field="${default_password}" expression="^1234$" break="never">
        <action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
        <action application="log" data="CRIT Open $${conf_dir}/vars.xml and change the default_password."/>
        <action application="log" data="CRIT Once changed type 'reloadxml' at the console."/>
        <action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
        <action application="sleep" data="10000"/>
      </condition>
    </extension>
 
    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="export" data="dialed_extension=$1"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
        <!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
 
    <extension name="echo">
      <condition field="destination_number" expression="^9196$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>
 
    <extension name="hold_music">
      <condition field="destination_number" expression="^9664$"/>
      <condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
        <action application="answer"/>
        <action application="execute_extension" data="is_secure XML features"/>
        <action application="playback" data="$${hold_music}"/>
        <anti-action application="set" data="zrtp_secure_media=true"/>
        <anti-action application="answer"/>
        <anti-action application="playback" data="silence_stream://2000"/>
        <anti-action application="execute_extension" data="is_zrtp_secure XML features"/>
        <anti-action application="playback" data="$${hold_music}"/>
      </condition>
    </extension>
 
    <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
 
  </context>

每个context中有多个extension，一个context中的extension与其他context中的extension在逻辑上是隔离的。

context的name用于在其它地方引用。

extension的name对呼叫流程没有任何影响。

extension的属性continue表示执行完action是否继续匹配其他extension，缺省为false。

extension中可以对一些condition（测试条件）进行判断，如果匹配测试条件所指定的表达式则执行对应的action（动作），如不匹配则执行对应的anti-action（反动作）。

condition可使用以下属性（可同时使用多个属性）：

field：通道变量。自定义变量需使用${}引起：
hour：小时，0~23。
mday：日，1~31。
minute：分，0~59。
minute-of-day：一天中的第几分钟，1~1440（1点等于60，中午等于720）。
mon：月，1~12。
mweek：本月中的第几周，1~6。
wday：一周中的第几天，1~7（周日为1）。可使用“-”连接两个值表示范围。
week：一年中的第几周，1~53。
year：当前的年，0~9999。
yday：一年中的第几天，1~366。

condition的break可为：

always；不管是否匹配，都停止。
never：不管是否匹配，都继续。
on-false：在匹配失败时停止（但继续处理其他extension），这是默认配置。
on-true：在匹配成功时停止（但会先完成对应action，然后继续处理其他的extension），不成功则继续匹配其他condition。

expression为field匹配的正则表达式。

没有field和expression的空condition被认为匹配所有规则，称其为绝对条件（Absolute Condition）。

action和anti-action有application属性，表示要执行的APP。

action和anti-action可有data属性，表示APP的参数，可用$N（从1开始）表示expression匹配的内容，用${API([ARG[ ...]])}表示API调用。如没有参数，也可省略。

action和anti-action将inline设置为true表示内联执行。

action可设置loop属性，如失败则重复执行直至指定次数。

可以使用sip_h_开头的通道变量添加扩展的SIP消息头。

默认号码如下（官方文档：https://freeswitch.org/confluence/display/FREESWITCH/Default+Dialplan+QRF）：

号码	说明
1000-1019	呼叫默认分机号（默认密码为1234）
2000-2002	呼叫组
30xx	电话会议，8kHz（其中xx可为00~99）
31xx	电话会议，宽带，16kHz（其中xx可为00~99）
32xx	电话会议，超宽带，32kHz（其中xx可为00~99）
33xx	电话会议，CD音质，48kHz（其中xx可为00~99）
4000	听取语音信箱
5000	示例IVR
5900	呼叫停泊
5901	呼叫取回
7243	组播
9178	收传真
9179	发传真
9180	铃音测试，使用远端生成的回铃音
9181	铃音测试，产生英式铃音
9182	铃音测试，使用音乐当铃音，彩铃
9183	先应答，然后发送英式铃音
9184	先应答，然后发送音乐铃音
9191	ClueCon
9192	在log中显示Channel信息
9193	录像
9194	播放录像
9195	echo，回音测试，延迟5秒
9196	echo，回音测试
9197	milliwatte extension，铃音生成
9198	TGML铃音生成示例
9664	保持音乐

dialplan/public.xml

公共Context，一般用于网关路由。这是文件内容的一部分：

   <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="transfer" data="$1 XML default"/>
      </condition>
    </extension>
 
    <X-PRE-PROCESS cmd="include" data="public/*.xml"/>
 
  </context>

大部分参数与dialplan/default.xml相同。

directory/

用户目录。支持多个域（Domain），每个域可以写到一个XML文件中。

directory/default.xml

用户目录默认域。这是文件内容的一部分：

   <domain name="$${domain}">
    <params>
      <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})},${verto_contact(${dialed_user}@${dialed_domain})}"/>
    </params>
 
    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>
 
    <groups>
      <group name="default">
        <users>
          <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
        </users>
      </group>
 
      <group name="sales">
        <users>
          <!--
              type="pointer" is a pointer so you can have the
              same user in multiple groups.  It basically means
              to keep searching for the user in the directory.
          -->
          <user id="1000" type="pointer"/>
          <user id="1001" type="pointer"/>
          <user id="1002" type="pointer"/>
          <user id="1003" type="pointer"/>
          <user id="1004" type="pointer"/>
        </users>
      </group>
 
    </groups>
  </domain>

domain的name属性为租户域的名字。

params定义了该域中所有用户的公共参数。dial-string参数为呼叫字符串。

variables定义了一些公共变量，在用户主叫或被叫时，这些变量会绑定到相应的Channel上形成Channel Variable。

还定义了很多组（group），组名并无特殊意义。组里面包含很多用户（user），可以是指向已存在用户的指针。使用组并不是必需的，可以将users直接放到domain下一级。

params和variables可以出现在user、group、domain中，当有重复时按此优先级从高到低。

directory/default/

本地用户目录。

directory/default/1000.xml

一个本地用户。文件内容如下：

 <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>

user_context的值为此用户发起的呼叫的拨号计划的context。

freeswitch.xml

主配置文件，使用X-PRE-PROCESS指令的include语句载入其他文件。这是文件内容的一部分：

 <?xml version="1.0"?>
<document type="freeswitch/xml">
  <X-PRE-PROCESS cmd="include" data="vars.xml"/>
 
  <section name="configuration" description="Various Configuration">
    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
  </section>
 
  <section name="dialplan" description="Regex/XML Dialplan">
    <X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
  </section>
 
  <section name="chatplan" description="Regex/XML Chatplan">
    <X-PRE-PROCESS cmd="include" data="chatplan/*.xml"/>
  </section>
 
  <section name="directory" description="User Directory">
    <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
  </section>
 
  <section name="languages" description="Language Management">
    <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/fr/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/ru/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/he/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/es/es_ES.xml"/>
    <X-NO-PRE-PROCESS cmd="include" data="lang/es/es_MX.xml"/>
    <X-PRE-PROCESS cmd="include" data="lang/pt/pt_BR.xml"/>
    <X-NO-PRE-PROCESS cmd="include" data="lang/pt/pt_PT.xml"/>
    <X-NO-PRE-PROCESS cmd="include" data="lang/sv/*.xml"/>
  </section>
</document>

ivr_menus/

IVR菜单配置。

ivr_menus/demo_ivr.xml

IVR菜单配置示例。文件内容如下：

 <include>
  <!-- demo IVR setup -->
  <!-- demo IVR, Main Menu -->
  <menu name="demo_ivr"
      greet-long="phrase:demo_ivr_main_menu"
      greet-short="phrase:demo_ivr_main_menu_short"
      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
      exit-sound="voicemail/vm-goodbye.wav"
      confirm-macro=""
      confirm-key=""
      tts-engine="flite"
      tts-voice="rms"
      confirm-attempts="3"
      timeout="10000"
      inter-digit-timeout="2000"
      max-failures="3"
      max-timeouts="3"
      digit-len="4">
 
    <!-- The following are the definitions for the digits the user dials -->
    <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
    <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/>
    <entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/>    <!-- FS echo -->
    <entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/>    <!-- MOH -->
    <entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/>    <!-- ClueCon -->
    <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>       <!-- Screaming monkeys -->
    <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>                  <!-- demo sub menu -->
    <!-- Using a regex in the digits tag lets you define a dial pattern for the caller
         You may define multiple regexes if you need a different pattern for some reason -->
    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
    <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
  </menu>
 
  <!-- Demo IVR, Sub Menu -->
  <menu name="demo_ivr_submenu"
      greet-long="phrase:demo_ivr_sub_menu"
      greet-short="phrase:demo_ivr_sub_menu_short"
      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
      exit-sound="voicemail/vm-goodbye.wav"
      timeout="15000"
      max-failures="3"
      max-timeouts="3">
 
    <!-- The demo IVR sub menu prompt basically just says, "press star to return to previous menu..." -->
    <entry action="menu-top" digits="*"/>
   </menu>
 
 
<!-- TTS sample; non-functional but it demonstrates say: and TTS -->
<!--
  <menu name="demo3"
      greet-long="say:Press 1 to join the conference, Press 2 to join the other conference"
      greet-short="say:Press 1 to join the conference, Press 2 to join the other conference"
      invalid-sound="say:invalid extension"
      exit-sound="say:exit sound"
      timeout ="15000"
      max-failures="3">
    <entry action="menu-exit" digits="*"/>
    <entry action="menu-play-sound" digits="1" param="say:You pressed 1"/>
    <entry action="menu-exec-app" digits="2" param="transfert 1000 XML default"/>
    <entry action="menu-exec-app" digits="3" param="transfert 1001 XML default"/>
  </menu>
-->
</include>

每个menu描述一个菜单，有以下属性（如文件为相对路径，则相对于/usr/local/freeswitch/sounds）：

digit-len：最大收号位数。
exit-sound：最后菜单退出时（一般是超时没有按键）的提示音文件。
greet-long：最开始的欢迎音文件。
greet-short：用户长时间没有按键的简短提示音文件。
inter-digit-timeout：两次按键的最大间隔（毫秒）。
invalid-sound：用户按键错误的提示音文件。
max-failures：用户按键错误的最多次数。
max-timeouts：最大超时次数。
name：菜单名。必需唯一，用于在其它地方引用。
timeout：超时时间（毫秒）。即多长时间没有收到按键就超时，播放其他提示音。

菜单有若干个菜单项entry，其action可为：

menu-exec-app：执行一个APP。
menu-sub：执行下级菜单。
menu-top：返回上级菜单。

lang

语言配置目录。

lang/en

英语配置目录。

lang/en/demo

英语配置示例。

lang/en/demo/demo-ivr.xml

英语IVR的Phrase宏示例。这是文件内容的一部分：

   <macro name="demo_ivr_main_menu" pause="100">
    <input pattern="(.*)">
      <match>
        <!-- string together several existing sound files to create one long greeting -->
        <action function="play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>
        <action function="play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>
        <action function="play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>
        <!-- note that you can do more than just play files, e.g. have pauses and do TTS -->
 
        <!-- Menu option 1: Call FreeSWITCH conference-->
        <action function="play-file" data="ivr/ivr-enter_ext_pound.wav"/>
        <action function="play-file" data="silence_stream://1500"/>
        <action function="play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/1.wav"/>
 
        <!-- Menu option 2: Do FreeSWITCH echo test -->
        <action function="play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/2.wav"/>
 
        <!-- Menu option 3: Listen to Music on Hold -->
        <action function="play-file" data="ivr/ivr-to_listen_to_moh.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/3.wav"/>
 
        <!-- Menu option 4: Register for ClueCon -->
        <action function="play-file" data="ivr/ivr-register_for_cluecon.wav"/>
        <action function="play-file" data="digits/4.wav"/>
 
        <!-- Menu option 5: Listen to screaming monkeys -->
        <action function="play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/5.wav"/>
 
        <!-- Menu option 6: Hear a sample submenu -->
        <action function="play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/6.wav"/>
 
        <!-- Menu option 9: Repeat these options -->
        <action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/9.wav"/>
        <action function="play-file" data="silence_stream://2000"/>
      </match>
    </input>
  </macro>

name为宏名称，用于其他地方引用。

pause表示每个action之间暂停多长时间（毫秒）。

可指定多个input，根据pattern正则表达式匹配不同的输入。

function可为：

play-file：播放文件。
say：使用Say接口放音。除了data属性，还需要type和method属性。
speak-text：使用TTS将文本内容放音。

data可使用$N引用pattern匹配的内容。

lang/en/en.xml

英语配置。这是文件内容的一部分：

 <include>
  <language name="en" say-module="en" sound-prefix="$${sound_prefix}" tts-engine="cepstral" tts-voice="callie">
    <phrases>
      <macros>
        <X-PRE-PROCESS cmd="include" data="demo/*.xml"/> <!-- Note: this now grabs whole subdir, previously grabbed only demo.xml -->
        <!-- voicemail_en_tts is purely implemented with tts, we have the files based one that is the default. -->
        <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/> <!-- vm/tts.xml if you want to use tts and have cepstral -->
        <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/> <!-- dir/tts.xml if you want to use tts and have cepstral -->
        <X-PRE-PROCESS cmd="include" data="ivr/*.xml"/>  <!-- IVR and custom phrases go here -->
      </macros>
      <X-PRE-PROCESS cmd="include" data="vm/voicemail_ivr.xml"/>
    </phrases>
  </language>
</include>

name为语言的名字，用于在其它地方引用。

say-module表示该语言用来支持Say接口的模块，本例中为mod_say_en。

sound-prefix为该语言Say接口的声音文件的路径前缀。

tts-engine为TTS引擎。

tts-voice为TTS嗓音。

sip_profiles

SIP Profile配置。

sip_profiles/external.xml

SIP external Profile配置。这是文件内容的一部分：

 <profile name="external">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>
 
  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>
 
  <settings>
    <param name="context" value="public"/>
    <param name="auth-calls" value="false"/>
  </settings>
</profile>

大部分参数与sip_profiles/internal.xml相同。

sip_profiles/external/

SIP external Profile配置目录。

sip_profiles/external/example.xml

SIP external Profile网关配置示例。文件内容如下：

 <include>
  <!--<gateway name="asterlink.com">-->
  <!--/// account username *required* ///-->
  <!--<param name="username" value="cluecon"/>-->
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <!--<param name="realm" value="asterlink.com"/>-->
  <!--/// username to use in from: *optional* same as  username, if blank ///-->
  <!--<param name="from-user" value="cluecon"/>-->
  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
  <!--<param name="from-domain" value="asterlink.com"/>-->
  <!--/// account password *required* ///-->
  <!--<param name="password" value="2007"/>-->
  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
  <!--<param name="extension" value="cluecon"/>-->
  <!--/// proxy host: *optional* same as realm, if blank ///-->
  <!--<param name="proxy" value="asterlink.com"/>-->
  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
  <!--<param name="register-proxy" value="mysbc.com"/>-->
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <!--<param name="expire-seconds" value="60"/>-->
  <!--/// do not register ///-->
  <!--<param name="register" value="false"/>-->
  <!-- which transport to use for register -->
  <!--<param name="register-transport" value="udp"/>-->
  <!--How many seconds before a retry when a failure or timeout occurs -->
  <!--<param name="retry-seconds" value="30"/>-->
  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
  <!--<param name="caller-id-in-from" value="false"/>-->
  <!--extra sip params to send in the contact-->
  <!--<param name="contact-params" value=""/>-->
  <!-- Put the extension in the contact -->
  <!--<param name="extension-in-contact" value="true"/>-->
  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
  <!--<param name="ping" value="25"/>-->
  <!--<param name="cid-type" value="rpid"/>-->
  <!--rfc5626 : Abilitazione rfc5626 ///-->
  <!--<param name="rfc-5626" value="true"/>-->
  <!--rfc5626 : extra sip params to send in the contact-->
  <!--<param name="reg-id" value="1"/>-->
  <!--</gateway>-->
</include>

各参数的含义如下：

caller-id-in-from：是否将主叫号码放到SIP的From首部。省略则为false，即放到Remote-Party-ID首部。
contact-params：SIP的Contact首部的额外参数。具体参数需根据实际情况而定。
expire-seconds：SIP的REGISTER消息的Expires首部。省略则为3600。单位为秒。
extension：来话中的分机号，即被叫号码。省略则取username。
from-domain：SIP的From首部的domain值。省略则取realm。
from-user：SIP的From首部的用户信息。省略则取username。
outbound-proxy：呼叫使用的代理服务器，即SIP的INVITE消息发送的地址。可与register-proxy不同。
password：密码。必需。
ping：发送SIP的OPTIONS消息的时间间隔。如果失败，则从该网关注销，并将其设置为down状态。
proxy：代理服务器。省略则取realm。
realm：SIP网关服务器地址。可为域名或IP，如果端口不是5060则需加上“:端口”。省略则取网关名，即gateway的name值。必需。
register：是否注册到网关。省略则为true。如果是中继，则不需要注册。
register-proxy：注册使用的代理服务器。省略则取realm。
register-transport：SIP消息的承载。可为：tcp、udp。
retry-seconds：注册失败或超时后，重新注册的等待秒数。
username：用户名，会出现在SIP消息的Authorization头中。必需。

sip_profiles/internal.xml

SIP internal Profile配置。这是文件内容的一部分：

 <profile name="internal">
  <gateways>
  </gateways>
 
  <domains>
    <domain name="all" alias="true" parse="false"/>
  </domains>
 
  <settings>
    <param name="context" value="public"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- port to bind to for sip traffic -->
    <param name="sip-port" value="$${internal_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="nat.auto"/>
  </settings>
</profile>

profile的name用于在其它地方引用。

param的name可为：

aggressive-nat-detection：对SIP包进行深度NAT检测，以决定使用那个IP地址。
apply-inbound-acl：SIP来话的ACL鉴权方式。
apply-nat-acl：处理SIP包NAT的ACL鉴权方式。nat.auto为使用RFC1918规定的私网地址并去掉本地网络的地址来替换Contact信息。
auth-calls：是否对来话进行鉴权。即所有从该Profile进来的INVITE请求都需要经过Digest验证。
codec-prefs：来话和去话支持的媒体编码列表。
context：来话进入Dialplan中哪个Context进行路由。本地用户设置的user_context的优先级更高。
dialplan：Dialplan类型。
disable-rtp-auto-adjust：是否禁用RTP自动调整。
dtmf-type：DTMF类型。inband为使用带内DTMF，info为使用SIP的INFO消息。
enable-100rel：是否在收到183消息时发送PRACK证实消息。
ext-rtp-ip：NAT环境中公网的RTP IP。该设置会影响SDP中的IP。可为：IP地址（直接指定公网IP）、autonat:IP地址（直接指定公网IP，更智能）、stun:STUN服务器地址（使用STUN服务器获取公网IP）、host:域名（通过DNS域名解析获取公网IP）、auto（自动检测公网IP）、auto-nat（如路由器支持NAT-PMP或uPnP，则自动使用这些协议获取公网IP）。
ext-sip-ip：NAT环境中公网的SIP IP。该设置会影响SDP中的IP。可为：IP地址（直接指定公网IP）、autonat:IP地址（直接指定公网IP，更智能）、stun:STUN服务器地址（使用STUN服务器获取公网IP）、host:域名（通过DNS域名解析获取公网IP）、auto（自动检测公网IP）、auto-nat（如路由器支持NAT-PMP或uPnP，则自动使用这些协议获取公网IP）。
force-register-db-domain：强制注册使用此域存储在数据库中。可删除。
force-register-domain：强制注册使用此域。可删除。
force-subscription-domain：强制订阅使用此域。可删除。
inbound-codec-prefs：来话支持的媒体编码列表。
inbound-bypass-media：来话是否启用媒体绕过模式。
inbound-late-negotiation：是否使用晚协商。
inbound-zrtp-passthru
manage-presence：是否启用列席。
manage-shared-appearence：是否启用共享线路呈现。
media-option：媒体选项。为resume-media-on-hold表示，如果FreeSWITCH处于媒体绕过状态，当话机按下Hold键时，FreeSWITCH将回到有媒体的状态。为bypass-media-after-att-xfer表示，如在执行出席转移前处于媒体绕过状态，转接执行时通过re-INVITE要回媒体，等到转接结束后再回到媒体绕过状态。
NDLB-force-rport：是否使用NDLB配置启用rport参数支持。
odbc-dsn：数据库ODBC的DSN。格式为：[odbc://]NAME:[USERNAME]:[PASSWORD]。
outbound-codec-prefs：去话支持的媒体编码列表。
rtp-ip：RTP的IP地址。
sip-capture：是否使用Homer方式进行抓包。
sip-ip：SIP的IP地址。
sip-port：该Profile监听的SIP端口。
track-calls：是否将通话的当前状态实时写到数据库中。

vars.xml

使用X-PRE-PROCESS指令的set语句定义一些全局变量。这是文件内容的一部分：

   <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="use_profile=external"/>

data各变量的含义如下：

api_on_startup：FreeSWITCH启动时执行的API及参数。
default_password：默认用户密码。
domain：租户域。默认是主机的IP地址。
domain_name：租户域的名字。用于Dialplan中。
global_codec_prefs：全局支持的媒体编码列表。
internal_sip_port：internal Profile监听的SIP端口。
outbound_codec_prefs：去话支持的媒体编码列表。
sound_prefix：声音文件的路径前缀。

另外，系统自动设置的变量有（系统自动设置的变量是在vars.xml加载前设置的，因而可以在vars.xml中覆盖它们）：

hostname：主机名。
local_ip_v4：本地IP（IPv4）。
local_mask_v4：本地掩码（IPv4）。
local_ip_v6：本地IP（IPv6）。
switch_serial：随机产生的序列号。
base_dir：FreeSWITCH的安装路径。
recordings_dir：录音路径，可以在启动时用-recordings选项指定。
sound_prefix：声音文件的路径前缀，默认是/usr/local/freeswitch/sounds/en/us/callie。
sounds_dir：声音文件的目录，默认是/usr/local/freeswitch/sounds。
core_uuid：自动生成的UUID字符串，用于标志本FreeSWITCH实例，每次启动都会改变。
zrtp_enabled：是否启用zrtp。
nat_public_addr：检测到的NAT公网地址。
nat_private_addr：NAT私网地址。
nat_type：NAT类型。

posted @ 2022-12-20 21:22 garvenc 阅读(4946) 评论(0) 编辑收藏举报

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	freeswitch.xml
	\|- vars.xml
	\|- autoload_configs/*.xml
	\| [autoload_configs/acl.conf.xml]
	\| [autoload_configs/callcenter.conf.xml]
	\| [autoload_configs/cdr_csv_conf.xml]
	\| [autoload_configs/cdr_pg_csv.conf.xml]
	\| [autoload_configs/conference.conf.xml]
	\| [autoload_configs/distributor.conf.xml]
	\| [autoload_configs/event_socket.conf.xml]
	\| [autoload_configs/fifo.conf.xml]
	\| [autoload_configs/ivr.conf.xml]
	\| `- ivr_menus/*.xml
	\| [ivr_menus/demo_ivr.xml]
	\| [autoload_configs/local_stream.conf.xml]
	\| [autoload_configs/lua.conf.xml]
	\| [autoload_configs/modules.conf.xml]
	\| [autoload_configs/nibblebill.conf.xml]
	\| [autoload_configs/post_load_modules.conf.xml]
	\| [autoload_configs/sofia.conf.xml]
	\| `- sip_profiles/*.xml
	\| [sip_profiles/external.xml]
	\| `- sip_profiles/external/*.xml
	\| [sip_profiles/external/example.xml]
	\| [sip_profiles/internal.xml]
	\| [autoload_configs/switch.conf.xml]
	\| [autoload_configs/tts_commandline.conf.xml]
	\| [autoload_configs/xml_cdr.conf.xml]
	\| [autoload_configs/xml_curl.conf.xml]
	\| [autoload_configs/xml_rpc.conf.xml]
	\|- dialplan/*.xml
	\| [dialplan/default.xml]
	\| `- dialplan/default/*.xml
	\| [dialplan/public.xml]
	\| `- dialplan/public/*.xml
	\|- directory/*.xml
	\| [directory/default.xml]
	\| `- directory/default/*.xml
	\| [directory/default/1000.xml]
	`- lang/en/*.xml
	[lang/en/en.xml]
	`- lang/en/demo/*.xml
	[lang/en/demo/demo-ivr.xml]

	<configuration name="acl.conf" description="Network Lists">
	<network-lists>
	<!--
	These ACL's are automatically created on startup.

	rfc1918.auto - RFC1918 Space
	nat.auto - RFC1918 Excluding your local lan.
	localnet.auto - ACL for your local lan.
	loopback.auto - ACL for your local lan.
	-->

	<list name="lan" default="allow">
	<node type="deny" cidr="192.168.42.0/24"/>
	<node type="allow" cidr="192.168.42.42/32"/>
	</list>

	<!--
	This will traverse the directory adding all users
	with the cidr= tag to this ACL, when this ACL matches
	the users variables and params apply as if they
	digest authenticated.
	-->
	<list name="domains" default="deny">
	<!-- domain= is special it scans the domain from the directory to build the ACL -->
	<node type="allow" domain="$${domain}"/>
	<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
	<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
	</list>

	</network-lists>
	</configuration>

	<queues>

	<queue name="support@default">
	<param name="strategy" value="longest-idle-agent"/>
	<param name="moh-sound" value="$${hold_music}"/>
	<!--<param name="record-template" value="$${recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
	<param name="time-base-score" value="system"/>
	<param name="max-wait-time" value="0"/>
	<param name="max-wait-time-with-no-agent" value="0"/>
	<param name="max-wait-time-with-no-agent-time-reached" value="5"/>
	<param name="tier-rules-apply" value="false"/>
	<param name="tier-rule-wait-second" value="300"/>
	<param name="tier-rule-wait-multiply-level" value="true"/>
	<param name="tier-rule-no-agent-no-wait" value="false"/>
	<param name="discard-abandoned-after" value="60"/>
	<param name="abandoned-resume-allowed" value="false"/>
	</queue>

	</queues>

	<agents>
	<!--<agent name="1000@default" type="callback" contact="[leg_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />-->
	</agents>
	<tiers>
	<!-- If no level or position is provided, they will default to 1. You should do this to keep db value on restart. -->
	<!-- <tier agent="1000@default" queue="support@default" level="1" position="1"/> -->
	</tiers>

	<configuration name="cdr_csv.conf" description="CDR CSV Format">
	<settings>
	<!-- 'cdr-csv' will always be appended to log-base -->
	<!--<param name="log-base" value="/var/log"/>-->
	<param name="default-template" value="example"/>
	<!-- This is like the info app but after the call is hung up -->
	<!--<param name="debug" value="true"/>-->
	<param name="rotate-on-hup" value="true"/>
	<!-- may be a b or ab -->
	<param name="legs" value="a"/>
	<!-- Only log in Master.csv -->
	<!-- <param name="master-file-only" value="true"/> -->
	</settings>
	<templates>
	<template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
	<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
	<template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
	<template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
	<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
	<template name="opencdrrate">"${uuid}","${signal_bond}","${direction}","${ani}","${destination_number}","${answer_stamp}","${end_stamp}","${billsec}","${accountcode}","${userfield}","${network_addr}","${regex('${original_caller_id_name}'\|^.)}","${sip_gateway_name}"</template>
	</templates>
	</configuration>

	<configuration name="cdr_pg_csv.conf" description="CDR PG CSV Format">
	<settings>
	<!-- See parameters for PQconnectdb() at http://www.postgresql.org/docs/8.4/static/libpq-connect.html -->
	<param name="db-info" value="host=localhost dbname=cdr connect_timeout=10" />
	<!-- CDR table name -->
	<!--<param name="db-table" value="cdr"/>-->

	<!-- Log a-leg (a), b-leg (b) or both (ab) -->
	<param name="legs" value="a"/>

	<!-- Directory in which to spool failed SQL inserts -->
	<!-- <param name="spool-dir" value="$${log_dir}/cdr-pg-csv"/> -->
	<!-- Disk spool format if DB connection/insert fails - csv (default) or sql -->
	<param name="spool-format" value="csv"/>
	<param name="rotate-on-hup" value="true"/>

	<!-- This is like the info app but after the call is hung up -->
	<!--<param name="debug" value="true"/>-->
	</settings>
	<schema>
	<field var="local_ip_v4"/>
	<field var="caller_id_name"/>
	<field var="caller_id_number"/>
	<field var="destination_number"/>
	<field var="context"/>
	<field var="start_stamp"/>
	<field var="answer_stamp"/>
	<field var="end_stamp"/>
	<field var="duration" quote="false"/>
	<field var="billsec" quote="false"/>
	<field var="hangup_cause"/>
	<field var="uuid"/>
	<field var="bleg_uuid"/>
	<field var="accountcode"/>
	<field var="read_codec"/>
	<field var="write_codec"/>
	<!-- <field var="sip_hangup_disposition"/> -->
	<!-- <field var="ani"/> -->
	</schema>
	</configuration>

	<caller-controls>
	<group name="default">
	<control action="mute" digits="0"/>
	<control action="deaf mute" digits="*"/>
	<control action="energy up" digits="9"/>
	<control action="energy equ" digits="8"/>
	<control action="energy dn" digits="7"/>
	<control action="vol talk up" digits="3"/>
	<control action="vol talk zero" digits="2"/>
	<control action="vol talk dn" digits="1"/>
	<control action="vol listen up" digits="6"/>
	<control action="vol listen zero" digits="5"/>
	<control action="vol listen dn" digits="4"/>
	<control action="hangup" digits="#"/>
	</group>
	</caller-controls>

	<profiles>
	<profile name="default">
	<param name="rate" value="8000"/>
	</profile>

	<profile name="wideband">
	</profile>

	<profile name="ultrawideband">
	</profile>

	<profile name="cdquality">
	</profile>
	</profiles>

	<configuration name="distributor.conf" description="Distributor Configuration">
	<lists>
	<!-- every 10 calls to test you will get foo1 once and foo2 9 times...yes NINE TIMES! -->
	<!-- this is not the same as 100 with 10 and 90 that would do foo1 10 times in a row then foo2 90 times in a row -->
	<list name="test">
	<node name="foo1" weight="1"/>
	<node name="foo2" weight="9"/>
	</list>
	</lists>
	</configuration>

	<configuration name="event_socket.conf" description="Socket Client">
	<settings>
	<param name="nat-map" value="false"/>
	<param name="listen-ip" value="::"/>
	<param name="listen-port" value="8021"/>
	<param name="password" value="ClueCon"/>
	<!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
	<!--<param name="stop-on-bind-error" value="true"/>-->
	</settings>
	</configuration>

	<configuration name="fifo.conf" description="FIFO Configuration">
	<settings>
	<param name="delete-all-outbound-member-on-startup" value="false"/>
	</settings>
	<fifos>
	<fifo name="cool_fifo@$${domain}" importance="0">
	<!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
	</fifo>
	</fifos>
	</configuration>

	<configuration name="ivr.conf" description="IVR menus">
	<menus>
	<X-PRE-PROCESS cmd="include" data="../ivr_menus/*.xml"/>
	</menus>
	</configuration>

	<configuration name="local_stream.conf" description="stream files from local dir">
	<!-- fallback to default if requested moh class isn't found -->
	<directory name="default" path="$${sounds_dir}/music/8000">
	<param name="rate" value="8000"/>
	<param name="shuffle" value="true"/>
	<param name="channels" value="1"/>
	<param name="interval" value="20"/>
	<param name="timer-name" value="soft"/>
	<!-- list of short files to break in with every so often -->
	<!--<param name="chime-list" value="file1.wav,file2.wav"/>-->
	<!-- frequency of break-in (seconds)-->
	<!--<param name="chime-freq" value="30"/>-->
	<!-- limit to how many seconds the file will play -->
	<!--<param name="chime-max" value="500"/>-->
	</directory>

	<directory name="moh/8000" path="$${sounds_dir}/music/8000">
	<param name="rate" value="8000"/>
	<param name="shuffle" value="true"/>
	<param name="channels" value="1"/>
	<param name="interval" value="20"/>
	<param name="timer-name" value="soft"/>
	</directory>
	</configuration>

	<configuration name="lua.conf" description="LUA Configuration">
	<settings>

	<!--
	Specify local directories that will be searched for LUA modules
	These entries will be pre-pended to the LUA_CPATH environment variable
	-->
	<!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
	<!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->

	<!--
	Specify local directories that will be searched for LUA scripts
	These entries will be pre-pended to the LUA_PATH environment variable
	-->
	<!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
	<!-- <param name="script-directory" value="$${script_dir}/?.lua"/> -->

	<!--<param name="xml-handler-script" value="/dp.lua"/>-->
	<!--<param name="xml-handler-bindings" value="dialplan"/>-->

	<!--
	The following options identifies a lua script that is launched
	at startup and may live forever in the background.
	You can define multiple lines, one for each script you
	need to run.
	-->
	<!--<param name="startup-script" value="startup_script_1.lua"/>-->
	<!--<param name="startup-script" value="startup_script_2.lua"/>-->

	<!--<hook event="CUSTOM" subclass="conference::maintenance" script="catch-event.lua"/>-->
	</settings>
	</configuration>

	<configuration name="modules.conf" description="Modules">
	<modules>
	<!-- Loggers (I'd load these first) -->
	<load module="mod_console"/>
	<!-- <load module="mod_graylog2"/> -->
	<load module="mod_logfile"/>
	<!-- <load module="mod_syslog"/> -->
	</modules>
	</configuration>

	<settings>
	<param name="odbc-dsn" value="bandwidth.com"/>
	</settings>

	<configuration name="sofia.conf" description="sofia Endpoint">

	<global_settings>
	<param name="log-level" value="0"/>
	<!-- <param name="abort-on-empty-external-ip" value="true"/> -->
	<!-- <param name="auto-restart" value="false"/> -->
	<param name="debug-presence" value="0"/>
	<!-- <param name="capture-server" value="udp:homer.domain.com:5060"/> -->

	<!--
	the new format for HEPv2/v3 and capture ID

	protocol:host:port;hep=2;capture_id=200;

	-->

	<!-- <param name="capture-server" value="udp:homer.domain.com:5060;hep=3;capture_id=100"/> -->
	</global_settings>

	<!--
	The rabbit hole goes deep. This includes all the
	profiles in the sip_profiles directory that is up
	one level from this directory.
	-->
	<profiles>
	<X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
	</profiles>

	</configuration>

	<configuration name="switch.conf" description="Core Configuration">

	<settings>
	<param name="max-sessions" value="1000"/>
	</settings>

	</configuration>

	<configuration name="tts_commandline.conf" description="TextToSpeech Commandline configuration">
	<settings>
	<!--
	Some variables will be replaced :
	${text}: input text (quoted)
	${rate}: sample rate (example: 8000)
	${voice}: voice_name passed to TTS(quoted)
	${file}: output file (quoted, including .wav extension)

	Example commands can be found at:
	https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Examplecommands
	-->
	<param name="command" value="echo ${text} \| text2wave -f ${rate} > ${file}"/>
	</settings>
	</configuration>

	<settings>
	<!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
	</settings>

	<configuration name="xml_curl.conf" description="cURL XML Gateway">
	<bindings>
	<binding name="example">
	<!-- <param name="gateway-url" value="http://www.freeswitch.org/gateway.xml" bindings="dialplan"/> -->
	</binding>
	</bindings>
	</configuration>

	<configuration name="xml_rpc.conf" description="XML RPC">
	<settings>
	<!-- The port where you want to run the http service (default 8080) -->
	<param name="http-port" value="8080"/>
	<!-- if all 3 of the following params exist all http traffic will require auth -->
	<param name="auth-realm" value="freeswitch"/>
	<param name="auth-user" value="freeswitch"/>
	<param name="auth-pass" value="works"/>


	<!-- regex pattern to match against commands called against this service.
	If a command with arguments matches, it will be logged at INFO level -->
	<!--<param name="commands-to-log" value=""/> -->

	</settings>
	</configuration>

garvenc

FreeSWITCH学习笔记：XML配置文件

加载顺序

autoload_configs/

autoload_configs/acl.conf.xml

autoload_configs/callcenter.conf.xml

autoload_configs/cdr_csv_conf.xml

autoload_configs/cdr_pg_csv.conf.xml

autoload_configs/conference.conf.xml

autoload_configs/distributor.conf.xml

autoload_configs/event_socket.conf.xml

autoload_configs/fifo.conf.xml

autoload_configs/ivr.conf.xml

autoload_configs/local_stream.conf.xml

autoload_configs/lua.conf.xml

autoload_configs/modules.conf.xml

autoload_configs/nibblebill.conf.xml

autoload_configs/post_load_modules.conf.xml

autoload_configs/sofia.conf.xml

autoload_configs/switch.conf.xml

autoload_configs/tts_commandline.conf.xml

autoload_configs/xml_cdr.conf.xml

autoload_configs/xml_curl.conf.xml

autoload_configs/xml_rpc.conf.xml

dialplan/

dialplan/default.xml

dialplan/public.xml

directory/

directory/default.xml

directory/default/

directory/default/1000.xml

freeswitch.xml

ivr_menus/

ivr_menus/demo_ivr.xml

lang

lang/en

lang/en/demo

lang/en/demo/demo-ivr.xml

lang/en/en.xml

sip_profiles

sip_profiles/external.xml

sip_profiles/external/

sip_profiles/external/example.xml

sip_profiles/internal.xml

vars.xml

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	<context name="default">

	<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="global" continue="true">
	<condition field="${call_debug}" expression="^true$" break="never">
	<action application="info"/>
	</condition>

	<condition field="${default_password}" expression="^1234$" break="never">
	<action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
	<action application="log" data="CRIT Open $${conf_dir}/vars.xml and change the default_password."/>
	<action application="log" data="CRIT Once changed type 'reloadxml' at the console."/>
	<action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
	<action application="sleep" data="10000"/>
	</condition>
	</extension>

	<extension name="Local_Extension">
	<condition field="destination_number" expression="^(10[01][0-9])$">
	<action application="export" data="dialed_extension=$1"/>
	<!-- bind_meta_app can have these args <key> [a\|b\|ab] [a\|b\|o\|s] <app> -->
	<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
	<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
	<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
	<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
	<action application="set" data="ringback=${us-ring}"/>
	<action application="set" data="transfer_ringback=$${hold_music}"/>
	<action application="set" data="call_timeout=30"/>
	<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
	<action application="set" data="hangup_after_bridge=true"/>
	<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
	<action application="set" data="continue_on_fail=true"/>
	<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
	<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
	<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
	<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
	<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
	<action application="answer"/>
	<action application="sleep" data="1000"/>
	<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
	</condition>
	</extension>

	<extension name="echo">
	<condition field="destination_number" expression="^9196$">
	<action application="answer"/>
	<action application="echo"/>
	</condition>
	</extension>

	<extension name="hold_music">
	<condition field="destination_number" expression="^9664$"/>
	<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32\|AES_CM_128_HMAC_SHA1_80)$">
	<action application="answer"/>
	<action application="execute_extension" data="is_secure XML features"/>
	<action application="playback" data="$${hold_music}"/>
	<anti-action application="set" data="zrtp_secure_media=true"/>
	<anti-action application="answer"/>
	<anti-action application="playback" data="silence_stream://2000"/>
	<anti-action application="execute_extension" data="is_zrtp_secure XML features"/>
	<anti-action application="playback" data="$${hold_music}"/>
	</condition>
	</extension>

	<X-PRE-PROCESS cmd="include" data="default/*.xml"/>

	</context>

	<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="transfer" data="$1 XML default"/>
	</condition>
	</extension>

	<X-PRE-PROCESS cmd="include" data="public/*.xml"/>

	</context>

	<domain name="$${domain}">
	<params>
	<param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})},${verto_contact(${dialed_user}@${dialed_domain})}"/>
	</params>

	<variables>
	<variable name="record_stereo" value="true"/>
	<variable name="default_gateway" value="$${default_provider}"/>
	<variable name="default_areacode" value="$${default_areacode}"/>
	<variable name="transfer_fallback_extension" value="operator"/>
	</variables>

	<groups>
	<group name="default">
	<users>
	<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
	</users>
	</group>

	<group name="sales">
	<users>
	<!--
	type="pointer" is a pointer so you can have the
	same user in multiple groups. It basically means
	to keep searching for the user in the directory.
	-->
	<user id="1000" type="pointer"/>
	<user id="1001" type="pointer"/>
	<user id="1002" type="pointer"/>
	<user id="1003" type="pointer"/>
	<user id="1004" type="pointer"/>
	</users>
	</group>

	</groups>
	</domain>

	<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>

	<?xml version="1.0"?>
	<document type="freeswitch/xml">
	<X-PRE-PROCESS cmd="include" data="vars.xml"/>

	<section name="configuration" description="Various Configuration">
	<X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
	</section>

	<section name="dialplan" description="Regex/XML Dialplan">
	<X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
	</section>

	<section name="chatplan" description="Regex/XML Chatplan">
	<X-PRE-PROCESS cmd="include" data="chatplan/*.xml"/>
	</section>

	<section name="directory" description="User Directory">
	<X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
	</section>

	<section name="languages" description="Language Management">
	<X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/fr/*.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/ru/*.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/he/*.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/es/es_ES.xml"/>
	<X-NO-PRE-PROCESS cmd="include" data="lang/es/es_MX.xml"/>
	<X-PRE-PROCESS cmd="include" data="lang/pt/pt_BR.xml"/>
	<X-NO-PRE-PROCESS cmd="include" data="lang/pt/pt_PT.xml"/>
	<X-NO-PRE-PROCESS cmd="include" data="lang/sv/*.xml"/>
	</section>
	</document>

	<include>
	<!-- demo IVR setup -->
	<!-- demo IVR, Main Menu -->
	<menu name="demo_ivr"
	greet-long="phrase:demo_ivr_main_menu"
	greet-short="phrase:demo_ivr_main_menu_short"
	invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
	exit-sound="voicemail/vm-goodbye.wav"
	confirm-macro=""
	confirm-key=""
	tts-engine="flite"
	tts-voice="rms"
	confirm-attempts="3"
	timeout="10000"
	inter-digit-timeout="2000"
	max-failures="3"
	max-timeouts="3"
	digit-len="4">

	<!-- The following are the definitions for the digits the user dials -->
	<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
	<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/>
	<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
	<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
	<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
	<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
	<entry action="menu-sub" digits="6" param="demo_ivr_submenu"/> <!-- demo sub menu -->
	<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
	You may define multiple regexes if you need a different pattern for some reason -->
	<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
	<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
	</menu>

	<!-- Demo IVR, Sub Menu -->
	<menu name="demo_ivr_submenu"
	greet-long="phrase:demo_ivr_sub_menu"
	greet-short="phrase:demo_ivr_sub_menu_short"
	invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
	exit-sound="voicemail/vm-goodbye.wav"
	timeout="15000"
	max-failures="3"
	max-timeouts="3">

	<!-- The demo IVR sub menu prompt basically just says, "press star to return to previous menu..." -->
	<entry action="menu-top" digits="*"/>
	</menu>


	<!-- TTS sample; non-functional but it demonstrates say: and TTS -->
	<!--
	<menu name="demo3"
	greet-long="say:Press 1 to join the conference, Press 2 to join the other conference"
	greet-short="say:Press 1 to join the conference, Press 2 to join the other conference"
	invalid-sound="say:invalid extension"
	exit-sound="say:exit sound"
	timeout ="15000"
	max-failures="3">
	<entry action="menu-exit" digits="*"/>
	<entry action="menu-play-sound" digits="1" param="say:You pressed 1"/>
	<entry action="menu-exec-app" digits="2" param="transfert 1000 XML default"/>
	<entry action="menu-exec-app" digits="3" param="transfert 1001 XML default"/>
	</menu>
	-->
	</include>

	<macro name="demo_ivr_main_menu" pause="100">
	<input pattern="(.*)">
	<match>
	<!-- string together several existing sound files to create one long greeting -->
	<action function="play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>
	<action function="play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>
	<action function="play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>
	<!-- note that you can do more than just play files, e.g. have pauses and do TTS -->

	<!-- Menu option 1: Call FreeSWITCH conference-->
	<action function="play-file" data="ivr/ivr-enter_ext_pound.wav"/>
	<action function="play-file" data="silence_stream://1500"/>
	<action function="play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/1.wav"/>

	<!-- Menu option 2: Do FreeSWITCH echo test -->
	<action function="play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/2.wav"/>

	<!-- Menu option 3: Listen to Music on Hold -->
	<action function="play-file" data="ivr/ivr-to_listen_to_moh.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/3.wav"/>

	<!-- Menu option 4: Register for ClueCon -->
	<action function="play-file" data="ivr/ivr-register_for_cluecon.wav"/>
	<action function="play-file" data="digits/4.wav"/>

	<!-- Menu option 5: Listen to screaming monkeys -->
	<action function="play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/5.wav"/>

	<!-- Menu option 6: Hear a sample submenu -->
	<action function="play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/6.wav"/>

	<!-- Menu option 9: Repeat these options -->
	<action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/>
	<action function="play-file" data="ivr/ivr-please.wav"/>
	<action function="play-file" data="voicemail/vm-press.wav"/>
	<action function="play-file" data="digits/9.wav"/>
	<action function="play-file" data="silence_stream://2000"/>
	</match>
	</input>
	</macro>

	<include>
	<language name="en" say-module="en" sound-prefix="$${sound_prefix}" tts-engine="cepstral" tts-voice="callie">
	<phrases>
	<macros>
	<X-PRE-PROCESS cmd="include" data="demo/*.xml"/> <!-- Note: this now grabs whole subdir, previously grabbed only demo.xml -->
	<!-- voicemail_en_tts is purely implemented with tts, we have the files based one that is the default. -->
	<X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/> <!-- vm/tts.xml if you want to use tts and have cepstral -->
	<X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/> <!-- dir/tts.xml if you want to use tts and have cepstral -->
	<X-PRE-PROCESS cmd="include" data="ivr/*.xml"/> <!-- IVR and custom phrases go here -->
	</macros>
	<X-PRE-PROCESS cmd="include" data="vm/voicemail_ivr.xml"/>
	</phrases>
	</language>
	</include>

	<profile name="external">
	<gateways>
	<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
	</gateways>

	<domains>
	<domain name="all" alias="false" parse="true"/>
	</domains>

	<settings>
	<param name="context" value="public"/>
	<param name="auth-calls" value="false"/>
	</settings>
	</profile>

	<include>
	<!--<gateway name="asterlink.com">-->
	<!--/// account username required ///-->
	<!--<param name="username" value="cluecon"/>-->
	<!--/// auth realm: optional same as gateway name, if blank ///-->
	<!--<param name="realm" value="asterlink.com"/>-->
	<!--/// username to use in from: optional same as username, if blank ///-->
	<!--<param name="from-user" value="cluecon"/>-->
	<!--/// domain to use in from: optional same as realm, if blank ///-->
	<!--<param name="from-domain" value="asterlink.com"/>-->
	<!--/// account password required ///-->
	<!--<param name="password" value="2007"/>-->
	<!--/// extension for inbound calls: optional same as username, if blank ///-->
	<!--<param name="extension" value="cluecon"/>-->
	<!--/// proxy host: optional same as realm, if blank ///-->
	<!--<param name="proxy" value="asterlink.com"/>-->
	<!--/// send register to this proxy: optional same as proxy, if blank ///-->
	<!--<param name="register-proxy" value="mysbc.com"/>-->
	<!--/// expire in seconds: optional 3600, if blank ///-->
	<!--<param name="expire-seconds" value="60"/>-->
	<!--/// do not register ///-->
	<!--<param name="register" value="false"/>-->
	<!-- which transport to use for register -->
	<!--<param name="register-transport" value="udp"/>-->
	<!--How many seconds before a retry when a failure or timeout occurs -->
	<!--<param name="retry-seconds" value="30"/>-->
	<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
	<!--<param name="caller-id-in-from" value="false"/>-->
	<!--extra sip params to send in the contact-->
	<!--<param name="contact-params" value=""/>-->
	<!-- Put the extension in the contact -->
	<!--<param name="extension-in-contact" value="true"/>-->
	<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
	<!--<param name="ping" value="25"/>-->
	<!--<param name="cid-type" value="rpid"/>-->
	<!--rfc5626 : Abilitazione rfc5626 ///-->
	<!--<param name="rfc-5626" value="true"/>-->
	<!--rfc5626 : extra sip params to send in the contact-->
	<!--<param name="reg-id" value="1"/>-->
	<!--</gateway>-->
	</include>