asterisk freeswitch 对比 学习

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Asterisk to FreeSWITCH Rosetta Stone

While FreeSWITCH is not a drop-in replacement for Asterisk, it does many of the same things that Asterisk does. This page is an attempt to help those familiar with Asterisk to leverage that knowledge and quickly locate that which is equivalent or analogous in FreeSWITCH. In most cases there isn't a direct, one-to-one translation, but rather similarities. For example, the "equivalent" of extensions.conf is (mostly) conf/dialplan/default.xml; but there are also features.xml, public.xml and /conf/dialplan/extensions/*xml.

  • If anyone has thoughts on this then by all means add what you know.

Configuration Files

Asterisk FreeSWITCH
extensions.conf conf/dialplan/default.xml; also features.xml, public.xml, extensions/*xml
logger.conf mod_console and mod_syslog
rtp.conf conf/autoload_configs/switch.conf.xml
sip.conf conf/directory/*.xml (see mod_sofia)
voicemail.conf mod_voicemail - voicemail.conf.xml, conf/directory/*xml
zapata.conf conf/autoload_configs/openzap.conf.xml
Realtime Use mod_xml_curl to fetch the user and/or dialplan in XML,mod_ldap for LDAP backend

Console Commands

Fs cli Asterisk Console FreeSWITCH
sip show peers/sip show registry sofia status profile internal
core set verbose 9 /log 7
core set debug 9 /debug 7
core show channels show channels / show calls
reload reloadxml
hangup request <channel> uuid_kill <uuid>
sip reload sofia profile internal rescan
sip set debug on sofia global siptrace on
sofia global debug (presence|sla|none)
sofia loglevel all [0-9]
sip set debug (ip|peer) sofia profile (internal|external) siptrace on
module load app_queue.so load mod_callcenter
core show uptime status
core show version version
console dial 1000 pa call 1000 (see mod_portaudio)

Miscellaneous

Asterisk FreeSWITCH
AMI mod_event_socket
asterisk -r fs_cli
asterisk -rx "command" fs_cli -x "command"
set verbose <verbosity> in CLI console loglevel 0-8 or console loglevel debug
chan_local Loopback
stop gracefully shutdown or ...

sip.conf params

Asterisk FreeSWITCH
dtmfmode In dialplan: start_dtmf

Asterisk experts: please add more information

Dialplan

Asterisk FreeSWITCH
exten => <extension></extension> tags
include => Misc._Dialplan_Tools_transfer
Realtime Mod xml curl to fetch the dialplan in XML
Answer Misc._Dialplan_Tools_answer
AGI Event Socket Outbound
Background Usually used for:
ChanSpy Misc._Dialplan_Tools_eavesdrop
Dial see bridge app
Dial(||L(x[:y][:z]) Limiting call time, use sched_hangup for the x and sched_broadcast for the:y and:z
Dial(SIP/${EXTEN}/sipuser) bridge with data="{sip_route_uri=sipuser}user/whatever" or data="sofia/whatever%domain.com^sipuser"
DumpChan Misc._Dialplan_Tools_info
Echo Misc._Dialplan_Tools_echo
Goto Misc._Dialplan_Tools_transfer
GotoIf Conditions in dialplan (<condition field="blah" expression="foo">)
Hangup Misc._Dialplan_Tools_hangup
Log Misc._Dialplan_Tools_log
Macro/GoSub Misc._Dialplan_Tools_execute_extension
MeetMe mod_conference
Monitor Misc._Dialplan_Tools_record_session
Monitor_exec Channel_Variables#api_hangup_hook
MP3Player mod_shout
Musiconhold mod_local_stream
NoCDR <action application="set" data="process_cdr=false"/>
NoOp Usually used for logging - Misc._Dialplan_Tools_log
Park Misc._Dialplan_Tools_park
Playback Misc._Dialplan_Tools_playback
Playtones Misc._Dialplan_Tools_gentones
Progress Misc._Dialplan_Tools_pre_answer
Queue mod_fifo
Read Misc._Dialplan_Tools_read
Record Misc._Dialplan_Tools_record
Set Misc._Dialplan_Tools_set
SetGlobal Misc._Dialplan_Tools_set_global
SIPGetHeader Auto set as variable - ${sip_h_HEADER} where HEADER is the header name
SIPAddHeader Set variable ${sip_h_HEADER} where HEADER is the header name you want to send
System Misc._Dialplan_Tools_system
Transfer Misc._Dialplan_Tools_redirect
Wait Misc._Dialplan_Tools_sleep
WaitExten Misc._Dialplan_Tools_play_and_get_digits

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