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: IVR play_and_get_digits
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