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