Rosetta Stone

• 1 Asterisk to FreeSWITCH Rosetta Stone
  • 1.1 Configuration Files
  • 1.2 Console Commands
  • 1.3 Miscellaneous
    • 1.3.1 sip.conf params
  • 1.4 Dialplan
  • 1.5 See Also

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

Asterisk Console	FreeSWITCH Fs cli	Shortcut Key
core show help	help	F1
core show uptime	status	F2
core show channels	show channels	F3
core show calls	show calls	F4
sip show settings	sofia statussofia status profile internal	F5F9
core reload	reloadxml	F6
core set verbose 0	/log 0	F7
core set verbose 9	/log 7	F8
core set debug 9	/debug 7
core show version	version	F12
channel originate sip/source extension destination	originate user/source destination xml default
channel originate sip/source application appname data	originate user/source &appname(data)
console dial 1000	pa call 1000 (see mod_portaudio)
database get family key	db select/family/key
dialplan show	xml_locate dialplan
hangup request channel	uuid_kill uuid
module load app_queue.so	load mod_callcenter
queue show	callcenter_config queue listcallcenter_config tier list
queue show queue_name	callcenter_config queue list agents queue_name@defaultcallcenter_config queue list members queue_name@default
sip reload	sofia profile internal rescan
sip set debug on	sofia global siptrace onsofia global debug (presence|sla	none)sofia loglevel all [0-9]
sip set debug (ip|peer)	sofia profile (internal|external) siptrace on
sip show peers	list_userssofia status profile internal reg
sip show peer sip_provider	sofia status gateway sip_provider

Miscellaneous

Asterisk	FreeSWITCH
AMI	mod_event_socket
asterisk -r	Command Line Interface (fs_cli)
asterisk -rx "command"	fs_cli -x "command"
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 Outboundivrd
Background	Usually used for:IVRplay_and_get_digits
ChanSpy	Misc._Dialplan_Tools_eavesdrop
DB	Mod_db
Dial	see bridge app
Dial(|	L(x[:y][: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">)
GotoIfTime	Conditions in dialplan with Misc._Dialplan_Tools_strftime or Time_of_Day_Routing
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_callcenter or 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

See Also

• Configuring FreeSWITCH
• Asterisk
• Convert Asterisk Dialplans to FreeSWITCH