存档九月 2018

FreeSWITCH核心命令

简介

下面的是根据最新的版本r14778(九月九号)中mod_commands模块提供的命令,这些命令可以使用方式有很多种,如下:

控制台

具体查看下面内容。 译者注:通过FreeSWITCH控制台使用

API/事件 接口

通过API或事件接口调用,如:

  • [[mod_event_socket]]
  • [[mod_xmpp_event]]
  • [[mod_erlang_event]]
  • [[mod_xml_rpc]]

脚本接口

通过脚本进行调用,如下:

  • [[mod_perl]]
  • [[mod_spidermonkey]]
  • [[mod_python]]
  • [[mod_lua]]

拨号方案调用

通过拨号方案进行调用,例子如下:

<source lang="xml">
 <extension name="Make API call from Dialplan">
   <condition field="destination_number" expression="^(999)$">
     <!-- next line calls hupall, so be careful! -->
     <action application="set" data="api_result=${hupall(normal_clearing)}"/>
   </condition>
 </extension>
</source>

其他例子:

<source lang="xml">
 <action application="set" data="api_result=${status()}"/>
 <action application="set" data="api_result=${version()}"/>
 <action application="set" data="api_result=${strftime()}"/>
 <action application="set" data="api_result=${expr(1+1)}"/>
</source>

如果API命令含有多个参数,通常都是以空格隔开。

<source lang="xml">
 <action application="set" data="api_result=${sched_api(+5 none avmd ${uuid} start)}"/>
</source>

API命令依赖于加载的相关模块,从每个模块注册的API命令中都能发现它们的踪影。

想要查看全部API命令列表的话,在cli中输入help或者show api即可。

注:如果你想从拨号方案中调用API命令的话,需要先确认拨号方案自带的dptools里面没有类似的命令。

核心命令

主要在http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_commands/mod_commands.c中实现。

注:一些状态或列表命令的返回结果默认是以逗号进行分隔的列表。一些模块的返回结果可能也会包含逗号,这样就导致针对结果的自动化处理比较困难。一个解决方法是,是在命令的最后加上“as xml”,这样返回的就是xml格式的结果。

acl

使用acl列表判断ip地址是否为合法访问。

Usage: acl

命令别名alias

别名:一种针对常用命令的快捷输入方式

用法: alias add <别名> <命令> | del [<别名>|*]

例子:

  freeswitch> alias add reloadall reloadacl reloadxml
  +OK
  freeswitch> alias add unreg sofia profile internal flush_inbound_reg
  +OK

别名在重启后需要重设,如果需要重启后仍然生效,需要使用stickyadd参数,如下:

  freeswitch> alias stickyadd reloadall reloadacl reloadxml
  +OK

注:只在mod_console中起作用,在fs_cli中无效。
译者注:mod_console为以前台模式启动的freeswitch的命令输入界面。而fs_cli指的是freeswitch的客户端。

bgapi

用于在线程中执行api命令

用法: bgapi <api命令>[ <参数>]

complete

Complete.

Usage: complete add <word>|del [<word>|*]

译者注:该命令从没用过,不知道干啥的,知道的童鞋,可以来更新该文档。

cond

运算指定的条件,并返回结果。

用法: cond <条件表达式> ? <true val> : <false val>

条件表达式支持的条件有:

== 等于
< 小于
> 大于

例子: 如果第一个值大于第二个,则返回true

 cond 5 > 3 ? true : false
 true

拨号方案中的例子:

   <action application="set" data="voicemail_authorized=${cond(${sip_authorized} == true ? true : false)}"/>

稍复杂的例子:

   <action application="set" data="voicemail_authorized=${cond(${sip_acl_authed_by} == domains ? false : ${cond(${sip_authorized} == true ? true : false)})}"/>

domain_exists

检查指定的domain是否存在:

 用法: domain_exists <domain>

eval

Eval (noop). 计算字符串,扩展通道变量.

用法: eval [uuid:<uuid> ]<expression>

例子:

 eval ${domain}
 10.15.0.94

 eval Hello, World!
 Hello, World!

 eval uuid:e72aff5c-6838-49a8-98fb-84c90ad840d9 ${channel-state}
 CS_EXECUTE

expand

执行变量扩展API。

 用法: [uuid:<uuid> ]<cmd> <args>

例子:

 expand originate sofia/internal/1001%${domain} 9999   

在这个例子中,扩展的变量是${domain}。比如domain的值是192.168.1.1,则扩展后执行的命令为:

 originate sofia/internal/1001%192.168.1.1 9999

fsctl

发送freeswitch控制消息。

 用法: fsctl [send_sighup |
               hupall |
               pause [inbound|outbound] |
               resume [inbound|outbound] |
               shutdown [cancel|elegant|asap|restart] |
               last_sps |
               sps [num] |
               sync_clock |
               sync_clock_when_idle |
               reclaim_mem |
               max_sessions |
               min_dtmf_duration [num] |
               max_dtmf_duration [num] |
               default_dtmf_duration [num] |
               loglevel [level] |
               verbose_events [on|off]
              ]

hupall

用于挂断呼向指定号码的通话。参数为:

clearing_type dialed_ext <extension number>

举个例子来说,杀掉正处于活跃状态、目标号码是1000的通话,命令为:

fsctl hupall normal_clearing dialed_ext 1000

sync_clock

FreeSWITCH不信任系统时间。当系统第一次启动的时候,从系统时间中获取样本时间,然后以此为基准使用单调时钟(monotonic clock)。你可以使用命令“fsctl sync_clock”将FreeSWITCH与系统时间进行同步。

注:该命令会立即生效,会影响CDR里面的时间统计。如会导致计费超前或延后,或者记录的挂断时间小于拨打时间。举个例子来说,如果FS的时钟比系统时间迟一个月,当进行时间同步后,CDR的呼叫记录里面就会出现有的呼叫持续时间为1个月。

命令fsctl sync_clock_when_idle要安全很多,作用和上面一样,但是要到系统中所有通道都空闲的时候才开始时间同步。这种方法不会对CDR产生影响。

sync_clock_when_idle

要到系统没有通话的时候才开始时间同步

sps

该设置会改变swithch.conf文件中设置的sessions-per-second(每秒并发通话数)属性限制

last_sps

查询显示目前生效的sessions-per-second属性。

pause

可以使用参数inbound或outbound来暂停创建呼入或呼出通话,如果没有指定参数的话,则呼入呼出都暂停。resume的用法类似。

min_dtmf_duration

例子:

fsctl min_dtmf_duration 800

译者注:没看懂,就不翻译出来误导人了!
This example sets the min_dtmf_duration switch parameter to 100ms. The number is in clock ticks where clockticks / 8 = ms. The min_dtmf_duration specifies the minimum DTMF duration to use on outgoing events. Events shorter than this will be increased in duration to match min_dtmf_duration. You cannot configure a DTMF duration on a profile that is less than this setting. You may increase this value, but cannot set it lower than 400 (the default). This value cannot exceed max_dtmf_duration. This setting can be changed in switch.conf.xml.

It is worth noting that many devices squelch in-band DTMF when sending RFC 2833. Devices that squelch in-band DTMF have a certain reaction time and clamping time which can sometimes reach as high as 40ms, though most can do it in less than 20ms. As the shortness of your DTMF event duration approaches this clamping threshold, the risk of your DTMF being ignored as a squelched event increases. If your call is always IP-IP the entire route, this is likely not a concern. However, when your call is sent to the PSTN, the RFC 2833 must be encoded in the audio stream. This means that other devices down the line (possibly a PBX or IVR you are calling into) might start considering your DTMF event a squelched tone and ignore it entirely. For this reason, it is recommended that you do not send DTMF events shorter than 80ms.

Checking the current value:

fsctl min_dtmf_duration 0

The code recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.

====max_dtmf_duration====

Example:

fsctl max_dtmf_duration 80000

This example sets the max_dtmf_duration switch parameter to 10,000ms (10 seconds). The number is in clock ticks (CT) where CT / 8 = ms. The max_dtmf_duration caps the playout of a DTMF event at the specified duration. Events exceeding this duration will be truncated to this duration. You cannot configure a duration on a profile that exceeds this setting. This setting can be lowered, but cannot exceed 192000 (the default). This setting cannot be set lower than min_dtmf_duration. This setting can be changed in switch.conf.xml.

Checking the current value:

fsctl max_dtmf_duration 0

The code recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.

====default_dtmf_duration====

Example:

fsctl default_dtmf_duration 2000

This example sets the default_dtmf_duration switch parameter to 250ms. The number is in clock ticks (CT) where CT / 8 = ms. The default_dtmf_duration specifies the DTMF duration to use on originated DTMF events or on events that are received without a duration specified. This value can be increased or lowered. This value is lower-bounded by min_dtmf_duration and upper-bounded by max_dtmf_duration. This setting can be changed in switch.conf.xml.

Checking the current value:

fsctl default_dtmf_duration 0

The code recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.

====verbose_events==== Enables verbose events. Verbose events have ”’every”’ channel variable in ”’every”’ event for a particular channel. Non-verbose events have only the pre-selected channel variables in the event headers. * This setting can also be set in [[switch.conf.xml]].

global_getvar

获取全局变量的值。如果没有提供参数,则返回所有全局变量的值。

用法: global_getvar <varname>

global_setvar

设置全局变量

用法: global_setvar <varname>=<value>

例子:

global_setvar foo=bar

group_call

返回组呼bridge字符串,组呼定义请参考[[XML User Directory Guide#Groups|call group]]。

Usage: group_call group@domain[+F][+A][+E]

+F将会以串行呼叫模式返回组成员(以“|”隔开各成员). +A将会以并行呼叫模式返回组成员(以“,”隔开各成员). +E将会议呼叫模式返回组成员(以:_:隔开各成员),关于企业呼叫请参考[[Freeswitch_IVR_Originate#Enterprise_originate|enterprise fashion]].

请注意:如果你需要设置在外呼通道上面设置用户变量,需要确保你的domain或被拨打组的变量列表里面没有设置dial-string和group-dial-string,用设置用户默认组里面的dial-string和group-dial-string来替代。这样的话,group_call将会返回user/101,user/将会设置你的外呼通道变量。

help

显示所有API命令的帮助信息。

用法: help

host_lookup

针对指定域名做主机查询(host lookup)。

用法: host_lookup <hostname>

hupall

断开现存通话。

用法: hupall <cause> [<variable> <value>]

挂断所有含有变量,并且值为的通话,挂机原因为。

例子:

originate {foo=bar}sofia/internal/someone1@server.com,sofia/internal/someone2@server.com &park

hupall normal_clearing foo bar

in_group

判断用户是否在指定的组中

用法: in_group <user>[@<domain>] <group_name>

is_lan_addr

判断IP是否为内网地址

用法: is_lan_addr <ip>

load

加载外部模块

用法: load <mod_name>

md5

返回指定数据的MD5值。

用法: md5 <data>

module_exists

检查模块是否存在。

用法: module_exists <module>

msleep

休眠指定毫秒

用法: msleep <休眠的毫秒数>

nat_map

用法: nat_map [status|reinit|republish] | [add|del] <port> [tcp|udp] [sticky] | [mapping] <enable|disable>
  • status – 用于显示NAT类型、外网IP(the external IP)以及当前映射的端口。
  • reinit – 重新初始化NAT模块。当你更换路由器或将路由器由NAT切换到UPnP模式的时候,使用该参数。
  • republish – 该参数会让FreeSWITCH重新(向路由器等)发布NAT映射信息。 正常情况下,没有必要使用该参数。
  • mapping – 该参数用于控制是否向NAT设备发送端口映射请求(可使用-nonatmap参数在系统启动时关闭该功能). 之所以存在该参数,是因为有可能需要通过NAT获取公网IP地址,而不需要通过NAT开启端口。

Note: sticky参数用于将映射信息固化下来,在下次FreeSWITCH重启后映射仍然生效。

警告: 如果你有多个网卡,并分别配置了使用相同端口的sip profiles。nat_map在映射端口的时候,会被弄昏头的,不需要将端口映射到哪个sip profile上面,千万别干这种挫事!

regex

执行正则表达式匹配。该参数会根据是否提供参数而实现不同的功能,如下:

  • 如果没提供该参数, ”regex” 将会执行正常的匹配,返回true或者false。
  • 如果提供该参数,如果匹配成功的话,会返回指定的子串。如果匹配失败,则返回全部源字符串。

默认的正则表达式分界符是|(管道符)。可以更改为~或者/,只要在字符串的前面加上’m:’。

Usage: regex <data>|<pattern>[|<subst string>]
       regex m:/<data>/<pattern>[/<subst string>]
       regex m:~<data>~<pattern>[~<subst string>]

例子:

 regex test1234|\d                  <== Returns "true"
 regex m:/test1234/\d               <== Returns "true"
 regex m:~test1234~\d               <== Returns "true"
 regex test|\d                      <== Returns "false"
 regex test1234|(\d+)|$1            <== Returns "1234"
 regex sip:foo@bar.baz|^sip:(.*)|$1 <== Returns "foo@bar.baz"
 regex testingonetwo|(\d+)|$1       <== Returns "testingonetwo" (no match)
 regex m:~30~/^(10|20|40)$/~$1      <== Returns "30" (no match)
 regex m:~30~/^(10|20|40)$/~$1~n    <== Returns "" (no match)
 regex m:~30~/^(10|20|40)$/~$1~b    <== Returns "false" (no match)

版本14727中的逻辑是,如果源字符串匹配匹配到结果,那么条件为false,但是这里仍有一个匹配结果,结果是1001。(这里的翻译是照字面翻译,小伙伴们,你们看懂了没?)
Logic in revision 14727 if the source string matches the result then the condition was false however there was a match and it is 1001.

regex 1001|(^\d{4}$)|$1
  • See also [[Regular_Expression]]

reload

重新加载模块。

用法: reload [-f] <mod_name>

reloadacl

重新加载ACL规则。

用法: reloadacl [reloadxml]

reloadxml

重新加载conf/freeswitch.xml的配置信息到内存中。

用法: reloadxml

show

输出多种(模块)状态报告。

 用法: show <item>
  item类型如下:
  codec
  endpoint
  application
  api
  dialplan
  file
  timer
  calls [count]
  channels [count|like <match string>]
  calls
  detailed_calls
  bridged_calls
  detailed_bridged_calls
  aliases
  complete
  chat
  management
  modules
  nat_map
  say
  interfaces
  interface_types
  tasks
  limits

XML格式输出: show foo as xml

修改输出分隔符: show foo as delim |

  • codec – 列出所有编码
  • endpoint – 列出所有endpoint类型模块
  • application – 列出所有应用程序
  • api – 列出所有api
  • dialplan – 列出拨号方案涉及的模块
  • file – 列出所有支持的文件类型
  • timer – 列出计时器timer模块
  • calls – 列出当前的通话[count]
  • channels – 列出当前的通道 [count|like ]
    注:关于calls与channels的对比,请参考Channels vs Calls
  • bridged_calls – 和”show calls”相同
  • detailed_calls – 和”show calls”类似,但是显示字段更多
  • detailed_bridged_calls – 和”show calls”类似,但是显示字段更多
  • aliases – 列出所有别名(别名干啥用的,暂时未查到)
  • complete – list command complete tables
  • chat – 列出所有chat模块,包括api、sms、conf等
  • management – list management?
  • modules – 列出所有模块
  • nat_map – 列出地址映射表
  • say – 列出有支持语言的say模块
  • interfaces – 列出所有接口
  • interface_types – 列出所有接口类型
  • tasks – 列出任务
  • registrations – 列出所有注册用户

Tips For Showing Calls and Channels

理解show calls/channels真义的最好方式是亲自去尝试。最近(2011.9)又在show命令家族中添加了几位:

  • show detailed_calls
  • show bridged_calls
  • show detailed_bridged_calls

这三个命令用于取代简单的”show calls”。
需要注意的是,”show detailed_calls”取代的是”show distinct_channels”。命令都是相似的,但是返回信息更多。
同样需要注意的是,这里并没有”show detailed_channels”命令,但是使用”show detailed_calls”会让你得到相同的结果。该命令能让你得到“单腿通话”(one-legged calls)或桥接后的通话信息,所以,少年,习惯这条新命令吧!

小贴士2: 有时,你需要获取某个特定的uuid,可以使用下面的方式。
假设你设置了通道变量presence_data,那可以使用下面的命令搜索符合条件的通道(即含有foo的通道): show channels like foo

like将会搜索下面的关键字段:

  • uuid
  • channel name
  • caller id name
  • caller id number
  • presence_data

注: presence_data 必须在bridgeoriginate期间设置,而不是在通道已经建立完成后才设置。

shutdown

停止FreeSWITCH程序。该命令只在cli中起作用,如果想作为api进行调用,需要使用fsctl shutdown

警告!在cli中运行shutdown会忽略掉参数,并立即退出!

用法: fsctl shutdown [cancel|elegant|asap|restart|now] 
  • cancel – 终止上一次提交的shutdown请求
  • elegant – 等待所有通话都停止后才关闭,允许新发起通话.
  • asap – 等待所有通话都停止后才关闭, 不再允许新通话.
  • restart – 在执行完“shutdown”后立即重启FreeSWITCH。
  • now – 立即重启FreeSWITCH。

当使用”elegant”, “asap”或者”now”参数后,还可以后跟restart命令,如下:

用法: fsctl shutdown [elegant|asap|now] restart

status

显示当前FreeSWITCH的运行状态

 用法: status

 freeswitch@internal> status
 UP 0 years, 0 days, 1 hour, 28 minutes, 4 seconds, 208 milliseconds, 305 microseconds
 FreeSWITCH is ready
 4 session(s) since startup
 0 session(s) 0/30                        <- 每秒创建的最大通话数 .. 来自switch.conf.xml
 1000 session(s) max                      <- 同时并存的最大通话数 .. 来自switch.conf.xml
 min idle cpu 0.00/100.00                 <- 达到拒接电话标准的最小闲置CPU值 .. 来自switch.conf.xml (如果该值被启动的话).

strftime_tz

根据不同的时区,显示格式化后的时间。需要查看linux时区标准列表的,请查看/usr/share/zoneinfo/zone.tab。

用法: strftime_tz <timezone> [format_string]

示例: strftime_tz US/Eastern %Y-%m-%d %T

unload

卸载外部模块

用法: unload [-f] <mod_name>

version

显示FreeSWITCH的版本号

用法: version [short]

xml_locate

xml_locate root: 返回FreeSWITCH使用的所有XML
xml_locate <section>: 返回指定<section>的XML

xml_locate directory
xml_locate configuration
xml_locate dialplan
xml_locate phrases

用法: xml_locate [root | <section> | <section> <tag> <tag_attr_name> <tag_attr_val>]

示例: xml_locate directory domain name example.com

xml_wrap

使用xml来包装API命令

用法: xml_wrap <command> <args>

呼叫管理命令

break

被废弃,请查看uuid_break命令

create_uuid

创建一个新的UUID,并以字符串的形式返回。

用法: create_uuid

originate

发起一个新的呼叫

Usage: originate <call_url> <exten>|&<application_name>(<app_args>) 
[<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]

参数:

  • <call_url> 呼叫目标URL.
    想多了解sofia sip URL语法的童鞋可以参考: [[Sofia|FreeSwitch Endpoint Sofia]]
  • 目标有如下几类:
    • 进入拨号方案进行路由的目标号码
    • &<application_name>(<app_args>)
      • “&” 表明后面跟的是应用名称,不是一个目标号码
      • (<app_args>) 可选参数 (不是所有应用都需要传递参数,比如park)
      • 下面是可以用在’&’后面的应用列表:
        park, bridge, javascript/lua/perl, playback (移除mod_native_file), and many others.
      • 注1: 用单引号传递含有空格的参数,如’&lua(test.lua arg1 arg2)’
      • 注2: 在&和application_name之间不能含有空格
  • 默认为’XML’,如果没有特别指定的话。
  • 默认为’default’,如果没有特别指定的话。
  • 主叫名称.
  • 主叫号码.
  • 超时时长(单位为秒).

可选参数:
这些可选参数使用大括号包裹,如:

originate {ignore\_early_media=true}sofia/example/user 8334

参数需要使用逗号隔开,例子如下:

originate {ignore_early_media=true,originate_timeout=2}sofia/example/user 8334
  • group_confirm_key
  • group_confirm_file
  • forked_dial
  • fail_on_single_reject
  • ignore_early_media
  • return_ring_ready
  • originate_retries
  • originate_retry_sleep_ms
  • origination_caller_id_name
  • origination_caller_id_number
  • originate_timeout
  • sip_auto_answer

更多变量,参考下面的地址:
[[Channel_Variables#Originate_related_variables|Description of originate’s related variables]]

例子:
假设,你想拨打一个本地注册的sip终端,号码为300,然后执行park操作,如下:
(注:本例中用的sip profile是example,你在实际测试的时候,需要改成你本地电话注册的sip profile,一般为internal)

originate sofia/example/300%pbx.internal &park()

又或者,你想将远程注册的sip终端连到拨号规则8600上

originate sofia/example/300@foo.com 8600

再或者,你想将远程注册的sip终端连到另一个远程终端

originate sofia/example/300@foo.com &bridge(sofia/example/400@bar.com)

还或者, 你甚至可以在接通后执行javascript脚本test.js

originate sofia/example/1000@somewhere.com &javascript(test.js)

如果运行的javascript脚本需要传递参数,则需要使用单引号括起来。

originate sofia/example/1000@somewhere.com '&javascript(test.js myArg1 myArg2)'

在发起呼叫前,设置通道变量

originate {ignore_early_media=true}sofia/mydomain.com/18005551212@1.2.3.4 15555551212

在发起呼叫期间,设置通道变量,并传递给另一个FS

originate {sip_h_X-varA=111,sip_h_X-varB=222}sofia/mydomain.com/18005551212@1.2.3.4 15555551212

注: 你可以设置任何类型的通道变量,即使是自定义变量。如果变量的值含有空格或逗号等符号,使用单引号括起来即可。

originate {my_own_var=my_value}sofia/mydomain.com/that.ext@1.2.3.4 15555551212
originate {my_own_var='my value'}sofia/mydomain.com/that.ext@1.2.3.4 15555551212

如果你想自造一段回铃音给被呼叫方听,try this:

originate {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/example/300@foo.com &bridge(sofia/example/400@bar.com)

如果你想发起呼叫后,通道进入”Ring-Ready”状态后就立即返回,try this:

originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full)

更多信息请查阅return ring ready

你可以将保持等待音乐设置为回铃音,if you want:

originate {ringback=\'/path/to/music.wav\'}sofia/gateway/name/number &bridge(sofia/gateway/name/othernumber)

你可以在后台发起一个呼叫(异步模式),播放一段60秒的提示消息:

bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message)

你可以指定被呼叫方的UUID,只需要下面几步:

  • 使用create_uuid创建一个UUID,待用。
  • 使用uuid_kill直接可以在对方未接听前杀掉该次呼叫。
  • 使用origination_uuid指定uuid之后,被叫方会在整个通话的生命周期中使用该UUID。 * originate {origination_uuid=…}user/100@domain.name.com

下面例子作用:发起一个到外部sip服务器echo conference的呼叫,然后转接到本地用户分机上

originate sofia/internal/9996@conference.freeswitch.org &bridge(user/105@default)

下面例子作用:向’default’以外的context上的分机发起呼叫(FreePBX会用到该特性,如context名字为context_1,context_2等等)

originate sofia/internal/2001@foo.com 3001 xml context_3

如果你想对多个分机发起呼叫,可以使用下面的命令:

originate user/1001,user/1002,user/1003 &park()

如果需要在收到early media的时候,将外呼的电话转入会议中,可以使用下面的两个命令,作用一样

originate sofia/example/300@foo.com &conference(conf_uuid-TEST_CON)
originate sofia/example/300@foo.com conference:conf_uuid-TEST_CON inline

   ( See [[Misc._Dialplan_Tools_InlineDialplan]] for more detail on 'inline' Dialplans )

下面的例子演示如何在A-leg上面使用loopback和inline
我是例子

pause

停止指定通道的媒体播放

用法: pause <uuid> <on|off>

uuid_answer

应答

用法: uuid_answer <uuid>
  • See Also: [[Misc.Dialplan_Tools_answer]]

uuid_audio

调整信道上面的音量,或直接通过一个媒体bug进行静音(读/写)

用法: uuid_audio <uuid> [start [read|write] [mute|level <level>]|stop]

level的值范围从-4到4,默认值为0。

uuid_break

断开发送至指定信道的媒体流。举例来说,如果此时正在信道上面播放一个音频文件,使用uuid_break命令,就会断开媒体,呼叫会顺着拨号方案、脚本等往下执行。

用法: uuid_break <uuid> [all]

如果使用all标记的话,所有信道上面正在排队等待播放的音频文件都会被移除,但是如果没有all标记的话,只有当前正在播放的音频文件会被断开。

uuid_bridge

桥接两条呼叫的腿。

Usage: uuid_bridge <uuid> <other_uuid>

uuid_bridge至少需要有一条腿是被呼通的。

uuid_broadcast

在一个指定UUID的信道上执行任意一个拨号方案程序。如果指定了某录音文件名,则代表将会在该信道上播放该文件。 执行拨号方案程序的语法规则是“app::args”。

用法: uuid_broadcast <uuid> <path> [aleg|bleg|both]

在选定的腿上执行应用程序,执行完毕后挂断,并指明挂机原因。

用法: uuid_broadcast <uuid> app[![hangup_cause]]::args [aleg|bleg|both]

具体应用举例如下:

 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e sorry.wav both
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say!::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say!user_busy::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::sorry.wav aleg

uuid_buglist

列出信道上面的媒体bug(media bugs)

用法: uuid_buglist <uuid>

uuid_chat

发送聊天信息

用法: <uuid> <text>

如果和会话(session,由uuid指定)相关的终端有一个receive_event handler,该消息会被发往终端,并以及时消息的形式显示出来。

uuid_debug_media

该命令过去为uuid_debug_audio,但是因为加入了一些视频的内容,所以改为现在的名字。

调试媒体流

用法:

<uuid> <read|write|both|vread|vwrite|vboth> <on|off>

使用“read”、“write”或者“both”(同时调试两个方向)作为语音流的方向,以进行调试。 在前面加上“v”,代表视频流的调试。

Read Format

“R %s b=%4ld %s:%u %s:%u %s:%u pt=%d ts=%u m=%d\n”

where the values are:

* switch_channel_get_name(switch_core_session_get_channel(session)),
* (long) bytes,
* my_host, switch_sockaddr_get_port(rtp_session->local_addr),
* old_host, rtp_session->remote_port,
* tx_host, switch_sockaddr_get_port(rtp_session->from_addr),
* rtp_session->recv_msg.header.pt, 
* ntohl(rtp_session->recv_msg.header.ts), 
* rtp_session->recv_msg.header.m

Write Format

“W %s b=%4ld %s:%u %s:%u %s:%u pt=%d ts=%u m=%d\n”

where the values are:

* switch_channel_get_name(switch_core_session_get_channel(session)),
* (long) bytes,
* my_host, switch_sockaddr_get_port(rtp_session->local_addr),
* old_host, rtp_session->remote_port,
* tx_host, switch_sockaddr_get_port(rtp_session->from_addr),
* send_msg->header.pt, 
* ntohl(send_msg->header.ts), 
* send_msg->header.m);

uuid_deflect

通过发送REFER方法,将当前FreeSWITCH上面的某个已经应答的sip呼叫转移走。

用法: uuid_deflect <uuid> <sip URL>

在命令执行后,uuid_deflect等待远端的应答,以此判断转移是否成功。远端返回的sip内容(sip fragment)将会作为uuid_deflect命令的返回结果。如果远端报告REFER成功,FreeSWITCH将会向那条信道发送bye信令。

举例如下:

uuid_deflect 0c9520c4-58e7-40c4-b7e3-819d72a98614 sip:info@example.net

返回内容:

Content-Type: api/response
Content-Length: 30

+OK:SIP/2.0 486 Busy Here

uuid_displace

将目标信道上面的语音流替换为指定的录音(文件)。

参数:

* uuid = 通话的唯一标识符(通过“show channels"可查看到)
* start|stop = 启动/停止该操作
* file = 要播放的语音源(wav,shout等等)路径
* limit = 语音替换(文件)的最大播放时长,秒数
* mux = 该选项将会导致原始的语音流与录音(文件)进行混音。比如,你在替换语音的时候,仍想与另一端进行会话(即在听到替换的录音文件的时候,也能听到对方的声音)。

用法:

uuid_displace <uuid> [start|stop] <file> [<limit>] [mux]

举例如下:

uuid_displace 1a152be6-2359-11dc-8f1e-4d36f239dfb5 start /sounds/test.wav 60
uuid_displace 1a152be6-2359-11dc-8f1e-4d36f239dfb5 stop /sounds/test.wav

uuid_display

更新话机的显示内容,前提是话机支持该功能。目前有Polycom和Snom等部分Sip话机支持该功能。

用法: <uuid> [<display>]

该命令会导致重新协商语音编码。SIP->RTP包的大小应该是0.020。如果在SPA系统话机上,设置为0.030的话,会引起DTMF延迟(DTMF lag)。当话机上的按键被按下的时候,我们可以通过fs_cli看到,但是会有4到6秒的延迟。

uuid_dual_transfer

将处于通话中的双方分别转移到不同的目的地。

-USAGE: <uuid> <A-dest-exten>[/<A-dialplan>][/<A-context>] <B-dest-exten>[/<B-dialplan>][/<B-context>]

uuid_dump

导出指定会话中的所有变量

Usage: uuid_dump <uuid> [format]

导出格式: XML

uuid_early_ok

停止忽略早期媒体(即正常播放early media)。 如果此时ignore_early_media=true,该命令将会停止忽略早期媒体(让参数ignore_early_media设置不起作用),并正常播放。

用法: uuid_early_ok

uuid_exists

检查给定的uuid是否存在。

用法: uuid_exists

uuid_flush_dtmf

刷新DTMF数字缓存,将在排队的DTMF全部送出

Usage: uuid_flush_dtmf

uuid_fileman

管理正在信道中播放的音频流,该音频来自一个语音文件。

用法: uuid_fileman <cmd:val>

命令如下:

*speed:<+[step]>|<-[step]>    语速
*volume:<+[step]>|<-[step]>   音量
*pause                         暂停
*stop                          停止
*truncate                      截断
*restart                      重启
*seek:<+[samples]>|<-[samples]> 定位

Samples,从字面上来讲,就是语音文件前进后退的取样数。在8KHZ的文件中,取样数8000代表的是一秒。同样,在16KHZ的文件中,16000代表的也是一秒。

uuid_getvar

获取指定的信道变量

用法: uuid_getvar

uuid_hold

保持通话

用法:

uuid_hold <uuid>           保持通话
uuid_hold off <uuid>       结束保持,恢复正常通话
uuid_hold toggle <uuid>    在保持和取消保持间切换

uuid_kill

重置(杀掉)指定的信道

用法: uuid_kill [cause]

uuid_limit

Apply or change limit(s) on a specified uuid.

Usage: uuid_limit [[/interval]] [number [dialplan [context]]]

See also [[Limit]]

uuid_media

Reinvite FreeSWITCH out of the media path:

Usage: uuid_media [off]

Reinvite FreeSWITCH back in:

Usage: uuid_media

uuid_media_reneg

API command to tell a channel to send a re-invite with optional list of new codecs

Usage: uuid_media_reneg

uuid_park

Park call

Usage: uuid_park

uuid_preanswer

Preanswer a channel.

Usage: uuid_preanswer

  • See Also: [[Misc._Dialplan_Tools_pre_answer]]

uuid_preprocess

Pre-process Channel

Usage: uuid_preprocess <>

uuid_recv_dtmf

Send DTMF digits to set.

Usage: uuid_recv_dtmf [@]

Use the character w for a .5 second delay and the character W for a 1 second delay.

Default tone duration is 2000ms .

uuid_send_dtmf

Send DTMF digits.

Usage: uuid_send_dtmf [@]

Use the character w for a .5 second delay and the character W for a 1 second delay.

Default tone duration is 2000ms .

uuid_send_info

Send info to the endpoint

Usage: uuid_send_info

uuid_session_heartbeat

Usage: uuid_session_heartbeat [sched] [0|]

uuid_setvar

Set a variable on a channel. If value is omitted, the variable is unset.

Usage: uuid_setvar [value]

uuid_setvar_multi

Set multiple vars on a channel.

Usage: uuid_setvar_multi =[;=[;…]]

uuid_simplify

This command directs FreeSWITCH to remove itself from the SIP signaling path if it can safely do so

Usage:

uuid_simplify

uuid_transfer

Transfers an existing call to a specific extension within a and . Dialplan may be “xml” or “directory”.

Usage:

uuid_transfer [-bleg|-both] [] []

The optional first argument will allow you to transfer both parties (-both) or only the party to whom is talking.(-bleg)

NOTE: if the call has been bridged, and you want to transfer either sides of the call, then you will need to use (or the API equivalent). If it’s not set, transfer doesn’t really work as you’d expect, and leaves calls in limbo.

Record/Playback Commands

uuid_record

Record the audio associated with the given UUID into a file. The start command causes FreeSWITCH to start mixing all call legs together and saves the result as a file in the format that the file’s extension dictates. (if available) The stop command will stop the recording and close the file. If media setup hasn’t yet happened, the file will contain silent audio until media is available. Audio will be recorded for calls that are parked. The recording will continue through the bridged call. If the call is set to return to park after the bridge, the bug will remain on the call, but no audio is recorded until the call is bridged again. (TODO: What if media doesn’t flow through FreeSWITCH? Will it re-INVITE first? Or do we just not get the audio in that case?)

Usage:

uuid_record [start|stop] []

Where limit is the max number of seconds to record.

If the path is not specified on start it will default to the channel variable “sound_prefix” or FreeSWITCH base_dir when the “sound_prefix” is empty.

You may also specify “all” for path when stop is used to remove all for this uuid

“stop” command must be followed by option.

[[Channel_Variables#Call_Recording_Related|See record’s related variables]]

Limit Commands

[[Limit#API|limit_reset]]

Reset a limit backend.

[[Limit#API|limit_status]]

Retrieve status from a limit backend.

[[Limit#API|limit_usage]]

Retrieve usage for a given resource.

[[Limit#API|uuid_limit_release]]

Manually decrease a resource usage by one.

[[Limit#API|limit_interval_reset]]

Reset the interval counter to zero prior to the start of the next interval.

Misc. Commands

bg_system

Execute a system command in the background.

Usage: bg_system

echo

Echo input back to the console echo This text will appear This text will appear

file_exists

Tests whether ”filename” exists.

file_exists filename

Examples:

file_exists /tmp/real_file true file_exists /tmp/missing_file false

Example dialplan usage:

 <extension name="play-news-announcements">
   <condition expression="${file_exists(${sounds_dir}/news.wav)}" expression="true"/>
     <action application="playback" data="${sounds_dir}/news.wav"/>
     <anti-action application="playback" data="${soufnds_dir}/no-news-is-good-news.wav"/>
   </condition>
 </extension>

”’Note”’ this tests whether FreeSWITCH can see the file, but the file may still be unreadable (permissions).

find_user_xml

Checks to see if a user exists; Matches user tags found in the directory, similar to [[user_exists]], but returns an XML representation of the user as defined in the directory (like the one shown in [[Mod_commands#user_exists|user_exists]]).

Usage: find_user_xml

Where key references a key specified in a directory’s user tag, user represents the value of the key, and the domain is the domain the user is assigned to.

list_users

Lists Users configured in Directory

Usage: list_users [group ] [domain ] [user ] [context ]

Example:

freeswitch@localhost> list_users group default

userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number
2000|default|192.168.20.73|default|sofia/internal/sip:2000@192.168.20.219:5060|techsupport|B#-Test 2000|2000
2001|default|192.168.20.73|default|sofia/internal/sip:2001@192.168.20.150:63412;rinstance=8e2c8b86809acf2a|techsupport|Test 2001|2001
2002|default|192.168.20.73|default|error/user_not_registered|techsupport|Test 2002|2002
2003|default|192.168.20.73|default|sofia/internal/sip:2003@192.168.20.149:5060|techsupport|Test 2003|2003
2004|default|192.168.20.73|default|error/user_not_registered|techsupport|Test 2004|2004

+OK

Search items can be combined:

freeswitch@localhost> list_users group default user 2004

userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number
2004|default|192.168.20.73|default|error/user_not_registered|techsupport|Test 2004|2004

+OK

sched_api

Schedule an API call in the future. Usage: sched_api [+@]

time is the UNIX timestamp at which the command should be executed. If it is prefixed by +, 

Scheduled task or group of tasks can be revoked with sched_del or unsched_api.

You could put “&” symbol at the end of the line to make command to be executed in its own thread.

Example: sched_api +1800 none originate sofia/internal/1000%${sip_profile} &echo() sched_api @600 check_sched log Periodic task is running…

sched_broadcast

Play a file to a specific call in the future. Usage: sched_broadcast [+]

Schedule execution of an application on a chosen leg(s) with optional hangup: Usage: sched_broadcast [+]

time is the UNIX timestamp at which the command should be executed (or if it is prefixed by +, the number of seconds to wait before executing the command)

Example: sched_broadcast +60 336889f2-1868-11de-81a9-3f4acc8e505e commercial.wav both sched_broadcast +60 336889f2-1868-11de-81a9-3f4acc8e505e say::en\snumber\spronounced\s12345 aleg

sched_del

Removes a prior scheduled group or task ID Usage: sched_del <group_name|task_id>

The one argument can either be a group of prior scheduled tasks or the returned task-id from sched_api.

Example: sched_del my_group sched_del 2

sched_hangup

Schedule a running call to hangup.

Usage: sched_hangup [+]

Note: sched_hangup +0 is the same as uuid_kill

sched_transfer

Schedule a transfer for a running call.

Usage: sched_transfer [+]

stun

Executes a STUN lookup. Usage: stun [:port]

Example: stun stun.freeswitch.org

system

Execute a system command.

Usage: system

The command is passed to the system shell, where it may be expanded or interpreted in ways you don’t expect. This can lead to security bugs if you’re not careful. For example, the following command is dangerous:

If a malicious remote caller somehow sets their caller ID name to “; rm -rf /”, you would unintentionally be executing this shell command:

log_caller_name; rm -rf /

time_test

Time test.

Usage: time_test [count]

Runs a test to see how bad timer jitter is. It runs the test count times (default 10) and tries to sleep for mss microseconds. It returns the actual timer duration along with an average.

Sample:

time_test 100 5

test 1 sleep 100 99
test 2 sleep 100 97
test 3 sleep 100 96
test 4 sleep 100 97
test 5 sleep 100 102
avg 98

timer_test

Timer test.

Usage: timer_test <10|20|40|60|120> [<1..200>] []

Runs a test to see how bad timer jitter is. Unlike time_test, this uses the actual freeswitch timer infrastructure to do the timer test and exercises the timers used for call processing.

First argument is the timer interval. Second is the count. Third is the timer name (“show timers” will give you a list)

Example:

timer_test 20 3

Avg: 16.408ms Total Time: 49.269ms

2010-01-29 12:01:15.504280 [CONSOLE] mod_commands.c:310 Timer Test: 1 sleep 20 9254
2010-01-29 12:01:15.524351 [CONSOLE] mod_commands.c:310 Timer Test: 2 sleep 20 20042
2010-01-29 12:01:15.544336 [CONSOLE] mod_commands.c:310 Timer Test: 3 sleep 20 19928

tone_detect

Start Tone Detection on a channel.

Usage: tone_detect [ ]

unsched_api

Unschedule an api command.

Usage: unsched_api

url_decode

Usage: url_decode

Url decode a string.

url_encode

Url encode a string.

Usage: url_encode

user_data

Retrieves user information (parameters or variables) as defined in the directory.

Usage: user_data @ [attr|var|param]

Where user is the user’s id, domain is the user’s domain, var|param specifies whether the info we’re requesting is a variable/parameter, and the name is the name (key) of the variable.

Example:

user_data 1000@192.168.1.101 param password

will return a result of 1234, and

user_data 1000@192.168.1.101 var accountcode

will return a result of 1000 from the example user shown in [[Mod_commands#user_exists|user_exists]], and

user_data 1000@192.168.1.101 attr id

will return the user’s actual alphanumeric ID (i.e. “john”) when number-alias=”1000″ was set as an attribute for that user.

user_exists

Checks to see if a user exists; Matches user tags found in the directory and returns either true/false:

Usage: user_exists

Where key references a key specified in a directory’s user tag, user represents the value of the key, and the domain is the domain the user is assigned to.

Example:

user_exists id 1000 192.168.1.101

will return true where there exists in the directory a user with a key called id whose value equals 1000: In the above example, we also could have tested for randomvar:

user_exists randomvar 45 192.168.1.101

And we would have received the same results, but:

user_exists accountcode 1000 192.168.1.101

or,

user_exists password 1000 192.168.1.101

Would have returned false.

See Also

  • [[Channel_Variables|Channel Variables]]

[[Category:Integration]] [[Category:Configuration]] [[Category:API]] [[Category:Modules]]