Docker内部からFreeSwitchのコマンドラインツール fs_cli を利用するため、event_socket.conf.xml で各種条件を設定します。

以下Socket Clientの設定で、FreeSwitchのコンテナ内で fs_cli のインターフェイスに入るためには、ホスト、ポート、パスワードオプションを付与します。

# fs_cli -H -P 8021 -p ClueCon


<configuration name="event_socket.conf" description="Socket Client">
    <param name="nat-map" value="false"/>
    <param name="listen-ip" value="$${local_ip_v4}"/>
    <param name="listen-port" value="8021"/>
    <param name="password" value="ClueCon"/>
    <param name="apply-inbound-acl" value="loopback.custom"/>
    <!--<param name="stop-on-bind-error" value="true"/>-->



Access Control List (ACL)-APICommands


reloadacl [<reloadxml>]

freeswitch@internal> reloadacl reloadxml


acl <ip> <list|net>

This command will allow you to test an IP address against one of your ACLs. Will return true or false. Use it to validate that your ACL behaves as expected. This test can also be a part of a dialplan <condition> test.

freeswitch@mybox> acl freeswitch@mybox> acl list_foo

Pre-defined ACLs

There are some ACLs automatically created on startup:

ACL name Description RFC 1918 Space RFC 1918, excluding your local LAN ACL for your local LAN ACL for your local LAN

fs_cli による設定ファイルのリロードコマンド

# fs_cli -H xx.xx.xx.xx -P 8021 -p password
freeswitch> reloadxml

Reload Command List

Item Reload Command Notes
Item Reload Command Notes
XML Dialplan reloadxml Run each time you edit XML dial file(s)
ACLs reloadacl Edit acl.conf.xml first
Voicemail reload mod_voicemail Edit voicemail.conf.xml first
Conference reload mod_conference Edit conference.conf.xml first
Add Sofia Gateway sofia profile <name> rescan Less intrusive - no calls dropped
Remove Sofia Gateway sofia profile <name> killgw <gateway_name> Less intrusive - no calls dropped
Restart Sofia Gateway sofia profile <name> killgw <gateway_name>
sofia profile rescan Less intrusive - no calls dropped
Add/remove Sofia Gateway sofia profile <name> restart More intrusive - all profile calls dropped
Local Stream see Mod_local_stream Edit localstream.conf.xml first
Update a lua file nothing necessary file is loaded from disk each time it is run
Update LCR SQL table nothing necessary SQL query is run for each new call
Update LCR options reload mod_lcr Edit lcr.conf.xml first
Update CID Lookup Options reload mod_cidlookup Edit cidlookup.conf.xml first
Update JSON CDR Options reload mod_json_cdr Edit json_cdr.conf.xml first
Update XML CDR Options reload mod_xml_cdr Edit xml_cdr.conf.xml first
Update XML CURL Server Response nothing, unless using cache

Ex) external-ipv6.xml を編集後再読込

freeswitch > sofia profile exxternal-ipv6 rescan

sofia help

freeswitch > sofia help
sofia global siptrace <on|off>
sofia        capture  <on|off>
             watchdog <on|off>

sofia profile <name> [start | stop | restart | rescan] [wait]
                     flush_inbound_reg [<call_id> | <[user]@domain>] [reboot]
                     check_sync [<call_id> | <[user]@domain>]
                     [register | unregister] [<gateway name> | all]
                     killgw <gateway name>
                     [stun-auto-disable | stun-enabled] [true | false]]
                     siptrace <on|off>
                     capture  <on|off>
                     watchdog <on|off>

sofia <status|xmlstatus> profile <name> [reg [<contact str>]] | [pres <pres str>] | [user <user@domain>]
sofia <status|xmlstatus> gateway <name>

sofia loglevel <all|default|tport|iptsec|nea|nta|nth_client|nth_server|nua|soa|sresolv|stun> [0-9]
sofia tracelevel <console|alert|crit|err|warning|notice|info|debug>

sofia help



Command Line Interface (fs_cli)




<configuration name="modules.conf" description="Modules">
    <!-- Loggers (I'd load these first) -->
    <load module="mod_console"/>

    <!-- Event Handlers -->
    <load module="mod_event_socket"/>

    <!-- Endpoints -->
    <load module="mod_sofia"/>

    <!-- Applications -->
    <load module="mod_commands"/>
    <load module="mod_conference"/>
    <load module="mod_dptools"/>

    <!-- Dialplan Interfaces -->
    <load module="mod_dialplan_xml"/>

    <!-- Codec Interfaces -->
    <load module="mod_spandsp"/>
    <load module="mod_opus"/>
    <load module="mod_opusfile"/>

    <!-- File Format Interfaces -->
    <load module="mod_sndfile"/>
    <load module="mod_native_file"/>



WebRTC (secure web socket wss settings)

edit /usr/local/freeswitch/conf/sip_profiles/internal.xml

# Set these params and save the file:

<param name="tls-cert-dir" value="/usr/local/freeswitch/certs" />

<param name="wss-binding" value=":7443" />

If behind N.A.T. make sure to set the ext-sip-ip and ext-rtp-ip in vars.xml to the public IP address of your FreeSWITCH.

If talking to clients both inside and outside the N.A.T. you must set the local-network-acl, and prefix the ext-sip-ip and ext-rtp-ip to autonat:X.X.X.X

Install Certificates

Layout of /usr/local/freeswitch/certs/wss.pem:


Cert, Key and Chain(s) are all contained in a single file in this order:

< cert >



< key >



< chain >



$ sudo cat cert.pem privkey.pem fullchain.pem > wss.pem


$ openssl s_client -host -port 7443


$ openssl s_client -help

Let’sEncryptの認証ファイルから作成した wss.pem は、SIPのTLS認証にも必要となりますがファイル名が異なります。wss.pem から内容をコピーして agent.pem, tls.pem, dtls-srtp.pem を作成します。


# make sure we have the correct permissions
chown root:ssl-cert my.certificate-privkey.pem
chown root:root my.certificate-cert.pem
chown root:root my.certificate-chain.pem
chown root:root my.certificate-fullchain.pem
chmod 640 my.certificate-privkey.pem
chmod 644 my.certificate-cert.pem
chmod 644 my.certificate-chain.pem
chmod 644 my.certificate-fullchain.pem

# copy file for webserver
cp my.certificate-fullchain.pem /etc/ssl/certs/my.certificate-fullchain.pem
cp my.certificate-privkey.pem /etc/ssl/private/my.certificate-privkey.pem

systemctl restart nginx

#setup freeswitch tls

#make sure the freeswitch directory exists
mkdir -p /etc/freeswitch/tls

#make sure the freeswitch certificate directory is empty
rm /etc/freeswitch/tls/*

#combine the certs into all.pem
cat my.certificate-fullchain.pem > /etc/freeswitch/tls/all.pem
cat my.certificate-privkey.pem >> /etc/freeswitch/tls/all.pem

#copy the certificates - not required, but useful to know what is in all.pem
cp my.certificate-cert.pem /etc/freeswitch/tls/cert.pem
cp my.certificate-chain.pem /etc/freeswitch/tls/chain.pem
cp my.certificate-fullchain.pem /etc/freeswitch/tls/fullchain.pem
cp my.certificate-privkey.pem /etc/freeswitch/tls/privkey.pem

#add symbolic links
ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/agent.pem
ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/tls.pem
ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/wss.pem
ln -s /etc/freeswitch/tls/all.pem /etc/freeswitch/tls/dtls-srtp.pem

#set the permissions
chown -R www-data:www-data /etc/freeswitch/tls

fs_cli -x "reload mod_sofia"


General NAT example scenarios

Scenario 1

FS ( -> NAT (Public IP -> INTERNET -> NAT (Public IP -> FS (

In this scenario you have to do two things.

  1. Setup a profile with ext-sip-ip and ext-rtp-ip set.
  2. If you setup ext-sip-ip and ext-rtp-ip on a profile then that profile CAN NOT BE USED BY ANYTHING BEHIND THE SAME NAT PERIOD.
  3. Port forward the ports for that profile.

Scenario 2

FS ( -> NAT (Public IP -> INTERNET -> NAT (Public IP -> PHONE (

Prior to setting up your plan you must decide which domain you are going to use for your phone registration and which port the profile will listen to. The general SIP structure being.

Phone {port} -> NAT 1 -> INTERNET -> NAT 2 -> {port}FS sip_profile {sip_profile domain} -> REGISTER {sip_profile domain} or Dialplan {sip_profile domain}

The easiest would be to use the default domain. Keeping in mind you still have to create a new sip_profile (for example, we’ll call it doublenat.xml) to handle the double NAT. The steps look like this.

  1. copy the external.xml and rename it doublenat.xml
  2. in doublenat.xml, change the profile name:
  3. add the
  4. add the
  5. add the
  6. add the
  7. change port to 5090 ( make sure to change it on the phone as well )
  8. add the
  9. add the
  10. register the phone
  11. check to see if your phone is registered with the doublenat profile.

sofia status profile doublenat

If your phone is not registered to doublenat then add :5090 to the domain/proxy on the phone configuration.

Example External IP:

Example Domain:

Keep in mind this profile will only be suitable for the double NAT scenario. Also keep in mind that this is the toughest scenario to conquer so you still may have adjustments on your phone or router.

Scenario 3

FS ( -> NAT (Public IP -> INTERNET -> Softphone Registering to FS

In this scenario you also need a dedicated profile. Getting devices to register to your FS instance sitting behind NAT.

  1. dedicated profile with ext-sip-ip and ext-rtp-ip set. Again NOTHING BEHIND THE SAME NAT WILL BE ABLE TO USE THIS PROFILE PERIOD.

Scenario 4

PHONE -> FS -> NAT -> Public Internet

In this case you setup a profile for the phones to register without the ext-sip-ip and ext-rtp-ip options set. This profile would be used for all devices registering to the

FS instance that are behind the same NAT.

Then your outbound call would traverse another profile setup to bust through the NAT with the proper ext-sip-ip and ext-rtp-ip options set.

Scenario 5

FS ( -> INTERNET -> NAT (Public IP -> Client (

More in depth article at Nat





     SIP and TLS settings.
     valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2
     default: tlsv1,tlsv1.1,tlsv1.2
  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>

:warning: nginxssl_protocols TLSv1 TLSv1.1 TLSv1.2; との整合性をチェックすること。




freeswitch@xxxxx> sofia status profile external
Name             	external
Domain Name      	N/A
Auto-NAT         	true
DBName           	sofia_reg_external
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	10.x.x.x
SIP-IP           	10.x.x.x
Ext-SIP-IP       	xx.xx.xx.xx
URL              	sip:[email protected]:5060
BIND-URL         	sip:[email protected]:5060;transport=udp,tcp
TLS-URL          	sip:[email protected]:5061
TLS-BIND-URL     	sips:[email protected]:5061;transport=tls
HOLD-MUSIC       	local_stream://moh
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	true
PROXY-MEDIA      	false
CALLS-IN         	0
CALLS-OUT        	0

保留音の設定 テストダイヤル 9664



<configuration name="local_stream.conf" description="stream files from local dir">
  <!-- fallback to default if requested moh class isn't found -->
  <directory name="default" path="$${sounds_dir}/music/8000">
    <param name="rate" value="8000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>
    <!-- list of short files to break in with every so often -->
    <!--<param name="chime-list" value="file1.wav,file2.wav"/>-->
    <!-- frequency of break-in (seconds)-->
    <!--<param name="chime-freq" value="30"/>-->
    <!-- limit to how many seconds the file will play -->
    <!--<param name="chime-max" value="500"/>-->

  <directory name="moh/8000" path="$${sounds_dir}/music/8000">
    <param name="rate" value="8000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>

  <directory name="moh/16000" path="$${sounds_dir}/music/16000">
    <param name="rate" value="16000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>

  <directory name="moh/32000" path="$${sounds_dir}/music/32000">
    <param name="rate" value="32000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="20"/>
    <param name="timer-name" value="soft"/>

  <directory name="moh/48000" path="$${sounds_dir}/music/48000">
    <param name="rate" value="48000"/>
    <param name="shuffle" value="true"/>
    <param name="channels" value="1"/>
    <param name="interval" value="10"/>
    <param name="timer-name" value="soft"/>



  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh/8000"/>



# make sounds-install moh-install


Test Calls

Test your new FreeSWITCH™ installation by configuring a pair of SIP phones and to place test calls and try out features in the default dialplan. If you’re already familiar with Asterisk, you could also start out by Connecting Freeswitch And Asterisk

Register two phones to FreeSWITCH with two different extension numbers.

Let’s assume for this example that you have your SIP phones registered as extensions 1000 and 1001.

The default config provides user directory files for extension numbers 1000 through 1019 with a registration password of 1234; the default voice-mail password is the same number as the extension, e.g. Extension 1000 has its voice-mail password set to 1000

Sample softphone configurations can be seen on the Softphones page.

Once you’ve done that, you can try out a bunch of features already setup in the test dialplan.

Basic Calling

  • Call between 1000 and 1001 by dialing the respective extension.
  • Call 4000 to login to your voicemail box.
  • 870 - Redial last called # from this extension.
  • *69 - Call back the last person to call this extension.
  • 8+[ext] - Place an intercom call to extension [ext]. If no answer, it does not go to voicemail but just keeps ringing. Some phones have additional functionality for intercom calls.

Other Tests

  • 9192 - Info test (prints information to the command line interface).fs_cliによるコンソール画面上にデバック情報が表示される。
  • 9193 - Video record test (records video to /tmp/testrecord.fsv)
  • 9194 - Video playback test (plays video from /tmp/testrecord.fsv)
  • 9195 - Echo test (5-second delay before returning echo).
  • 9196 - Echo test (real-time).
  • 9197 - Play milliwatt test tone.
  • 9198 - Tetris, synthesized via tone streaming.
  • 9664 - Test Music On Hold out (if you installed sound files).

Interactive Voice Response

  • 5000 - IVR demo (if you installed sound files).



1000 - 1019 Local extensions
** + extension number Intercepts a ringing phone (that is, call pickup)
2000 Samples call group: Sales
2001 Samples call group: Support
2002 Samples call group: Billing
3000 - 3399 Samples conference rooms
4000 or *98 Retrieves voicemail
5000 Demo IVR
5900 FIFO queue park
5901 FIFO queue retrieve
6000 Valet park/retrieval, manual
6001-6099 Valet park/retrieval, automatic
7243 RTP multicast page
0911 Group intercom example #1
0912 Group intercom example #2
0913 Emergency outbound conference example
9178 Example fax receive
9179 Example fax transmit
9180 Ring test, far end generates ring tone
9181 Ring test, send U.K. ring tone
9182 Ring test, send music as ring tone
9183 Answer, then send U.K. ring tone
9184 Answer, then send music as ring tone
9191 ClueCon registration
9192 Information dump
9195 Delayed echo test
9196 Echo test
9197 Milliwatt tone (test signal quality)
9198 Tetris
9664 Music on hold








# cd /usr/local/freeswitch/conf/directory/default
# cp 1000.xml 1100.xml


<user id="1100">
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="1100"/>
<variable name="toll_allow"
<variable name="accountcode" value="1100"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Gwen"/>
<variable name="effective_caller_id_number"
<variable name="outbound_caller_id_name"
<variable name="outbound_caller_id_number"
<variable name="callgroup" value="techsupport"/>


<extension name="Local_Extension">
<condition field="destination_number"


freeswitch> reloadxml




1000:受け手は、番号4000または*98と自身のID: 1000# とパスワード: 1000# の入力で、ボイスメールを聞くことが出来ます。

Main menu:
1—Listen to new messages
2—Listen to saved messages
5—Options menu (recorded name, greeting, and so on)
#—Exit voicemail
While listening to a message:
1—Replay message from the beginning
2—Save message
After listening to a message:
1—Replay message from the beginning
2—Save message
4—Send to e-mail (requires configuration)
7—Delete message





<group name="default">
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
<group name="sales">
	      type="pointer" is a pointer so you can have the
	      same user in multiple groups.  It basically means
	      to keep searching for the user in the directory.
<user id="1000" type="pointer"/>
<user id="1001" type="pointer"/>
<user id="1002" type="pointer"/>
<user id="1003" type="pointer"/>
<user id="1004" type="pointer"/>
<group name="billing">
<user id="1005" type="pointer"/>
<user id="1006" type="pointer"/>
<user id="1007" type="pointer"/>
<user id="1008" type="pointer"/>
<user id="1009" type="pointer"/>
<group name="support">
<user id="1010" type="pointer"/>
<user id="1011" type="pointer"/>
<user id="1012" type="pointer"/>
<user id="1013" type="pointer"/>
<user id="1014" type="pointer"/>

<group name="custom">
<user id="1000" type="pointer"/>
<user id="1100" type="pointer"/>


<extension name="group_dial_sales">
<condition field="destination_number" expression="^2000$">
<action application="bridge" data="${group_call(sales@${domain_name})}"/>
<extension name="group_dial_support">
<condition field="destination_number" expression="^2001$">
<action application="bridge" data="group/support@${domain_name}"/>
<extension name="group_dial_billing">
<condition field="destination_number" expression="^2002$">
<action application="bridge" data="group/billing@${domain_name}"/>
<extension name="group_dial_custom">
<condition field="destination_number" expression="^2003$">
<action application="bridge"


freeswitch@internal> reloadxml
freeswitch@internal> group_call custom
[email protected]]error/user_not_registered,[sip_invite_
domain=,[email protected]]sofia/internal/
sip:[email protected]:5060;rinstance=8eecf059256b51f1;fs_



Auto Nat

autonat and external ip

If you know your external ip, you can set it on your sofia profile and also instruct FreeSWITCH™ to set it when building the SDP:

< param name = "ext-sip-ip" value = "autonat:${external_sip_ip}" />

If you want to use the same profile to talk with things inside and outside your network you need to instruct FreeSWITCH™ when it should use the local IP and the external IP. The autonat: prefix toggles on the usage of the local-network-acl, if you prefix the IP like that it will activate the dynamic ability to tell when it should use ext-rtp-ip vs rtp-ip based on the acl match.


Sofia Configuration Files


This is the IP behind which FreeSWITCH is seen from the Internet, so if FreeSWITCH is behind NAT, this is basically the public IP that should be used for RTP.

ext-rtp-ip value Description Example
$${variable} Any variable from vars.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
w.x.y.z Specific numeric IP address <param name="ext-rtp-ip" value=""/>
autonat:w.x.y.z Used for LAN and WAN to avoid errors in the SIP CONTACT sent to LAN devices <param name="ext-rtp-ip" value="autonat:"/>
auto Guessed IP (guessed by looking in the IP routing table which interface is the default route) <param name="ext-rtp-ip" value="auto"/>
auto-nat FS uses uPNP or NAT-PMP to discover the public IP address it should use <param name="ext-rtp-ip" value="auto-nat"/>
stun:DNS name or IP address of STUN server FS uses the STUN server of your choice to discover the public IP address <param name="ext-rtp-ip" value=""/>
host:DNS name FS resolves the DNS name as the public IP address, so you can use a dynamic DNS host <param name="ext-rtp-ip" value=""/>

params with the ‘ext–’ prefix cited above when populated with yet-to-be-resolved DNS strings (e.g. name="ext–sip–ip" value="" or name="ext‑rtp–ip" value="" ) are resolved to IP addresses once only at FreeSWITCH load time and remain constant thereafter. FS is blind to (unaware of) any subsequent changes in your environment’s IP address. Thus, these ext– vars may become functionally incompatible with the environment’s current IP addresses with unspecified results in call flow at the network layer. FS restart is required for FS to capture the now-current, working IP address(es).



When receiving a REGISTER or INVITE, enable NAT mode automatically if IP address in Contact header matches an entry defined in the RFC 1918 access list. “acl” is a misnomer in this case because access will not be denied if the user’s contact IP doesn’t match.

<param name="apply-nat-acl" value="rfc1918"/>


These settings deal with authentication: requirements for identifying SIP endpoints to FreeSWITCH.


Users in the directory can have “auth-acl” parameters applied to them so as to restrict users access to a predefined ACL or a CIDR.

<param name="auth-calls" value="$${internal_auth_calls}"/>

Value can be “false” to disable authentication on this profile, meaning that when calls come in the profile will not send an auth challenge to the caller.


Please make sure to read SIP TLS before enabling certain features below as they may not behave as expected.


TLS: disabled by default, set to “true” to enable

<param name="tls" value="$${internal_ssl_enable}"/>


disabled by default, when enabled prevents sofia from listening on the unencrypted port for this connection. This can stop many generic brute force scripts and if all your clients connect over TLS then can help decrease the exposure of your FreeSWITCH server to the world.

<param name="tls-only" value="false"/>


additional bind parameters for TLS

<param name="tls-bind-params" value="transport=tls"/>


Port to listen on for TLS requests. (5061 will be used if unspecified)

<param name="tls-sip-port" value="$${internal_tls_port}"/>


Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server)

<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>


TLS version (“sslv2”, “sslv3”, “sslv23”, “tlsv1”, “tlsv1.1”, “tlsv1.2”). NOTE: Phones may not work with TLSv1

<param name="tls-version" value="$${sip_tls_version}"/>

When not set defaults to: “tlsv1,tlsv1.1,tlsv1.2”


If your agent.pem is protected by a passphrase stick the passphrase here to enable FreeSWITCH to decrypt the key.

<param name="tls-passphrase" value=""/>


If the client/server certificate should have the date on it validated to ensure it is not expired and is currently active.

<param name="tls-verify-date" value="true"/>


This controls what, if any security checks are done against server/client certificates. Verification is generally checking certificates are valid against the cafile.pem. Set to ‘in’ to only verify incoming connections, ‘out’ to only verify outgoing connections, ‘all’ to verify all connections, also ‘subjects_in’, ‘subjects_out’ and ‘subjects_all’ for subject validation (subject validation for outgoing connections is against the hostname/ip connecting to). Multiple policies can be split with a ‘|’ pipe, for example ‘subjects_in|subjects_out’. Defaults to none.

<param name="tls-verify-policy" value="none"/>


When certificate validation is enabled (tls-verify-policy) how deep should we try to verify a certificate up the chain again the cafile.pem file. By default only depth of 2.

<param name="tls-verify-depth" value="2"/>


If subject validation is enabled for incoming connections (tls-verify-policy set to ‘subjects_in’ or ‘subjects_all’) this is the list of subjects that are allowed (delimit with a ‘|’ pipe), note this only effects incoming connections for outgoing connections subjects are always checked against hostnames/ips.

<param name="tls-verify-in-subjects" value=""/>


External Profile

PSTN (public switched telephone network)既存電話網への接続サービスを提供している外部SIPプロバイダに登録することで、安価に外線通話・内線へのダイレクト通話が可能になります。
注) 通話料は安くなりますが、ほぼ全てのプロバイダが有料で、法人向けのサービス内容です。

設定例 port:5080


<gateway name=”“>
<param name=”username” value=”(YOUR-FREELYCALL-ACCOUNT-NUMBER)”/>
<param name=”realm” value=”[](“/>
<param name=”from-domain” value=”[](“/>
<param name=”password” value=”+++++++++++”/>
<param name=”register” value=”true”/>
<param name=”retry-seconds” value=”30″/>

fs_cliコンソールで、sofia sipの再起動と設定ファイルのリロードを実行します。

freeswitch> sofia profile external restart reloadxml

external::freelycall gateway sip:[email protected] REGED

freeswitch> sofia status
                     Name	   Type	                                      Data	State
            external-ipv6	profile	                  sip:mod_sofia@[::1]:5080	RUNNING (0)
            external-ipv6	profile	                  sip:mod_sofia@[::1]:5081	RUNNING (0) (TLS)
  	  alias	                                  internal	ALIASED
                 external	profile	          sip:[email protected]:5080	RUNNING (0)
                 external	profile	          sip:[email protected]:5081	RUNNING (0) (TLS)	gateway	                   sip:[email protected]	NOREG
     external::freelycall	gateway	                 sip:[email protected]	REGED
            internal-ipv6	profile	                  sip:mod_sofia@[::1]:5060	RUNNING (0)
                 internal	profile	          sip:[email protected]:5060	RUNNING (0)



<extension name=" freelycall-outbound">
<condition field="destination_number"
<action application="bridge"



<extension name="freelycall-inbound">
<condition field="destination_number"
<action application="set"
<action application="bridge" data="1000 XML default"/>


freeswitch> reloadxml



Sofia SIPコマンド

Reloading profiles and gateways

You can reload a specific SIP profile by issuing a rescan/restart command from the console

freeswitch> sofia profile <profile_name> [<rescan>|<restart>] reloadxml

freeswitch> sofia profile internal restart reloadxml
freeswitch> sofia profile external rescan reloadxml

View SIP Registrations

You can view all the devices that have registered by running the following from the console.

freeswitch> sofia status profile <profile name> reg

freeswitch> sofia status profile internal reg 
freeswitch> sofia status profile external reg

Configuring FreeSWITCH to use databases



<!--  <param name="core-db-dsn" value="pgsql://hostaddr= dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE'" />  -->
<!--  <param name="core-db-dsn" value="postgresql://freeswitch:@" />  -->
<param name="core-db-dsn" value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;" /> 
<!--  <param name="core-db-dsn" value="dsn:username:password" />  -->



<param name="odbc-dsn" value="odbc://your_dsn_name:your_db_user:your_db_password"/>


Perl-compatible regular expressions (PCRE)

Pattern Meaning
123 Match any string containing the sequence “123”
^123 Match any string beginning with the sequence “123”
123$ Match any string ending with the sequence “123”
^123$ Match any string that is exactly the sequence “123”
\d Match any single digit (0-9)
\d\d Match two consecutive digits
^\d\d\d$ Match any string that is exactly three digits long
^\d{7}$ Match any string that is exactly seven digits long
^(\d{7})$ Match any string that is exactly seven digits long, and store the matched value in a special variable named $1
^1?(\d{10})$ Matching any string that optionally begins with the digit “1” and contains an additional ten digits; store the ten digits in $1
^(3\d\d\d)$ Match any four-digit string that begins with the digit “3”, and store the matched value in $1

Regular Expression