Rosetta Stone

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Contents 1 Asterisk to FreeSWITCH Rosetta Stone 1.1 Configuration Files 1.2 Dialplan 1.3 SIP Configuration 1.4 Console Commands 1.5 Miscellaneous 1.6 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
sip.conf	conf/directory/*xml
voicemail.conf	mod_voicemail - voicemail.conf.xml, conf/directory/*xml
zapata.conf	conf/autoload_configs/openzap.conf.xml

Dialplan

Asterisk	FreeSWITCH
exten =>	<extension></extension> tags
include =>	Misc._Dialplan_Tools_transfer
Realtime	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
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
NoOp	Usually used for logging - Misc._Dialplan_Tools_log
Park	Misc._Dialplan_Tools_park
Playback	Misc._Dialplan_Tools_playback
Playtones	Misc._Dialplan_Tools_gentones
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

For an example on converting an Asterisk dialplan to a FreeSWITCH one see Converting Asterisk Dialplans to FreeSWITCH

SIP Configuration

Asterisk	FreeSWITCH
sip.conf	mod_sofia
Realtime	mod_xml_curl to fetch the user and/or dialplan in XML, mod_ldap for LDAP backend

Console Commands

Asterisk Console	FreeSWITCH Fs cli
sip show peers/sip show registry	sofia status profile internal
set verbose	/debug
show channels	show channels
reload	reloadxml
soft hangup <channel>	uuid_kill <uuid>
sip reload	sofia profile internal rescan
sip debug	sofia profile internal siptrace on (svn rev 12478)

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 ...

See Also

• Getting Started Guide
• Converting Asterisk to FreeSWITCH