contrib: Add Dockerfile to build and configure a FreeSWITCH

Rhizomatica is using FreeSWITCH and we should have an easy way to
test against it. A docker container with exposed ports seems like
the easiest. FreeSWITCH by default is giving us some exmaple numbers:

	* 5000 a menu... that allows DTMF
	* 9195 an echo test
	* 9198 tetris.

The config is copied on top of the default/big config that is
installed. If this PBX should be reached from the outside one needs
to change 127.0.0.1 to the external address and maybe configure the
acl as well to add more CIDRs.

Besides that
	make container
	make run

Will build it and start the container. Takes a bit of time and requires
docker. With it configure one can see things like:

2017-03-05 15:32:49.913912 [INFO] switch_channel.c:515 RECV DTMF 3:2000
2017-03-05 15:32:50.952752 [INFO] switch_channel.c:515 RECV DTMF 2:2000

Now to test DTMF in the system.

Change-Id: I7f3aa8c81b9e8698df090a05d2e41a41b67d8e3c
changes/70/1970/1
Holger Hans Peter Freyther 6 years ago
parent 42b073a233
commit 7166d0f448
  1. 25
      contrib/testpbx/Dockerfile
  2. 12
      contrib/testpbx/Makefile
  3. 29
      contrib/testpbx/README
  4. 34
      contrib/testpbx/configs/acl.conf.xml
  5. 832
      contrib/testpbx/configs/default.xml
  6. 422
      contrib/testpbx/configs/internal.xml
  7. 68
      contrib/testpbx/configs/public.xml
  8. 181
      contrib/testpbx/configs/switch.conf.xml
  9. 450
      contrib/testpbx/configs/vars.xml

@ -0,0 +1,25 @@
FROM debian:jessie
RUN apt-get update
RUN DEBIAN_FRONTEND=noninteractive apt-get install -y --no-install-recommends wget
# They use comodo.. it was hacked.. so don't bother trying to
# install the right root certificates...
RUN wget --no-check-certificate -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -
RUN echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
RUN apt-get update && apt-get install -y freeswitch-meta-all
# Change the config...
COPY configs/vars.xml /etc/freeswitch/vars.xml
COPY configs/acl.conf.xml /etc/freeswitch/autoload_configs/acl.conf.xml
COPY configs/switch.conf.xml /etc/freeswitch/autoload_configs/switch.conf.xml
COPY configs/public.xml /etc/freeswitch/dialplan/public.xml
COPY configs/default.xml /etc/freeswitch/dialplan/default.xml
COPY configs/internal.xml /etc/freeswitch/sip_profiles/internal.xml
# Prepare to run
# Reduce the number of ports.. as otherwise we wait a long time
EXPOSE 6000-6020/udp
EXPOSE 5060/udp
CMD /usr/bin/freeswitch -nf

@ -0,0 +1,12 @@
all: container
container:
docker build -t osmo-freeswitch-pbx:latest .
run:
docker run -it --name=osmo-freeswitch-pbx \
-p 5060:5060/udp -p 6000-6020:6000-6020/udp \
--rm=true osmo-freeswitch-pbx:latest
stop:
docker rm -f osmo-freeswitch-pbx

@ -0,0 +1,29 @@
Provide a semi-stable remote PBX system.
There is no preferred PBX but YaTE is pretty small and still
functional enough. Anyway Rhizomatica is using FreeSWITCH so
let's use that for testing.
This is creating a docker image with a SIP configuration that
will allow to record audio, have a DTMF menu using some fixed
numbers. Feel free to extend it to support bidirectional calls
and routing.
It is using the Debian packages and installs everything as I
am not interested to track dependencies and see what is missing.
Again feel free to optimize the size.
Build:
make
or
docker build -t yourimagename:tag .
Run:
docker run yourimagename:tag
SIP is exposed on 5060 of your port and audio on 6000-6020

@ -0,0 +1,34 @@
<configuration name="acl.conf" description="Network Lists">
<network-lists>
<!--
These ACL's are automatically created on startup.
rfc1918.auto - RFC1918 Space
nat.auto - RFC1918 Excluding your local lan.
localnet.auto - ACL for your local lan.
loopback.auto - ACL for your local lan.
-->
<list name="lan" default="allow">
<node type="allow" cidr="192.168.0.0/16"/>
</list>
<!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
-->
<list name="domains" default="allow">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="$${domain}"/>
<node type="allow" cidr="0.0.0.0/0"/>
<node type="allow" cidr="172.0.0.0/8"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
<node type="allow" cidr="192.168.0.0/16"/>
<node type="allow" cidr="10.0.0.0/16"/>
</list>
</network-lists>
</configuration>

@ -0,0 +1,832 @@
<?xml version="1.0" encoding="utf-8"?>
<!--
NOTICE:
This context is usually accessed via authenticated callers on the sip profile on port 5060
or transfered callers from the public context which arrived via the sip profile on port 5080.
Authenticated users will use the user_context variable on the user to determine what context
they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml
will build the domains ACL using this value.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="default">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!-- Example of doing things based on time of day.
year = 4 digit year. Example year="2009"
yday = 1-365
mon = 1-12
mday = 1-31
week = 1-52
mweek= 1-6
wday = 1-7
hour = 0-23
minute = 0-59
minute-of-day = 1-1440
Example:
<condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
do something ...
</condition>
-->
<extension name="tod_example" continue="true">
<condition wday="2-6" hour="9-18">
<action application="set" data="open=true"/>
</condition>
</extension>
<!-- Example of routing based on holidays
This example covers all US Federal holidays except for inauguration day.
-->
<extension name="holiday_example" continue="true">
<condition mday="1" mon="1">
<!-- new year's day -->
<action application="set" data="open=false"/>
</condition>
<condition wday="2" mweek="3" mon="1">
<!-- martin luther king day is the 3rd monday in january -->
<action application="set" data="open=false"/>
</condition>
<condition wday="2" mweek="3" mon="2">
<!-- president's day is the 3rd monday in february -->
<action application="set" data="open=false"/>
</condition>
<condition wday="2" mon="5" mday="25-31">
<!-- memorial day is the last monday in may (the only monday between the 25th and the 31st) -->
<action application="set" data="open=false"/>
</condition>
<condition mday="4" mon="7">
<!-- independence day -->
<action application="set" data="open=false"/>
</condition>
<condition wday="2" mday="1-7" mon="9">
<!-- labor day is the 1st monday in september (the only monday between the 1st and the 7th) -->
<action application="set" data="open=false"/>
</condition>
<condition wday="2" mweek="2" mon="10">
<!-- columbus day is the 2nd monday in october -->
<action application="set" data="open=false"/>
</condition>
<condition mday="11" mon="11">
<!-- veteran's day -->
<action application="set" data="open=false"/>
</condition>
<condition wday="5-6" mweek="4" mon="11">
<!-- thanksgiving is the 4th thursday in november and usually there's an extension for black friday -->
<action application="set" data="open=false"/>
</condition>
<condition mday="25" mon="12">
<!-- Christmas -->
<action application="set" data="open=false"/>
</condition>
</extension>
<extension name="global-intercept">
<condition field="destination_number" expression="^886$">
<action application="answer"/>
<action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/global)}"/>
<action application="sleep" data="2000"/>
</condition>
</extension>
<extension name="group-intercept">
<condition field="destination_number" expression="^\*8$">
<action application="answer"/>
<action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/${callgroup})}"/>
<action application="sleep" data="2000"/>
</condition>
</extension>
<extension name="intercept-ext">
<condition field="destination_number" expression="^\*\*(\d+)$">
<action application="answer"/>
<action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
<action application="sleep" data="2000"/>
</condition>
</extension>
<extension name="redial">
<condition field="destination_number" expression="^(redial|870)$">
<action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/>
</condition>
</extension>
<extension name="global" continue="true">
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
<condition field="${default_password}" expression="^1234$" break="never">
<action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
<action application="log" data="CRIT Open $${conf_dir}/vars.xml and change the default_password."/>
<action application="log" data="CRIT Once changed type 'reloadxml' at the console."/>
<action application="log" data="CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING "/>
<!-- <action application="sleep" data="10000"/> -->
</condition>
<!--
This is an example of how to auto detect if telephone-event is missing and activate inband detection
-->
<!--
<condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
<action application="set" data="rtp_payload_number=$1"/>
<anti-action application="start_dtmf"/>
</condition>
-->
<condition field="${rtp_has_crypto}" expression="^($${rtp_sdes_suites})$" break="never">
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
<!--
Since we have inbound-late-negotation on by default now the
above behavior isn't the same so you have to do one extra step.
-->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/>
<condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never">
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
<condition>
<action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/>
<action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
</condition>
</extension>
<!-- If sip_req_host is not a local domain then this has to be an external sip uri -->
<!--
<extension name="external_sip_uri" continue="true">
<condition field="source" expression="mod_sofia"/>
<condition field="${outside_call}" expression="^$"/>
<condition field="${domain_exists(${sip_req_host})}" expression="true">
<anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/>
</condition>
</extension>
-->
<!--
Snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference
-->
<extension name="snom-demo-2">
<condition field="destination_number" expression="^9001$">
<action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/>
<action application="transfer" data="3000"/>
</condition>
</extension>
<extension name="snom-demo-1">
<condition field="destination_number" expression="^9000$">
<!--<key> <light> <label> <user> <host> <profile> <action_name> <action>-->
<action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/>
<action application="playback" data="$${hold_music}"/>
</condition>
</extension>
<extension name="eavesdrop">
<condition field="destination_number" expression="^88(\d{4})$|^\*0(.*)$">
<action application="answer"/>
<action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1$2)}"/>
</condition>
</extension>
<extension name="eavesdrop">
<condition field="destination_number" expression="^779$">
<action application="answer"/>
<action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/>
<action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/>
<action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/>
<action application="eavesdrop" data="all"/>
</condition>
</extension>
<extension name="call_return">
<condition field="destination_number" expression="^\*69$|^869$|^lcr$">
<action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/>
</condition>
</extension>
<extension name="del-group">
<condition field="destination_number" expression="^80(\d{2})$">
<action application="answer"/>
<action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/>
<action application="gentones" data="%(1000, 0, 320)"/>
</condition>
</extension>
<extension name="add-group">
<condition field="destination_number" expression="^81(\d{2})$">
<action application="answer"/>
<action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/>
<action application="gentones" data="%(1000, 0, 640)"/>
</condition>
</extension>
<extension name="call-group-simo">
<condition field="destination_number" expression="^82(\d{2})$">
<action application="bridge" data="{leg_timeout=15,ignore_early_media=true}${group(call:$1@${domain_name})}"/>
</condition>
</extension>
<extension name="call-group-order">
<condition field="destination_number" expression="^83(\d{2})$">
<action application="bridge" data="{leg_timeout=15,ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/>
</condition>
</extension>
<extension name="extension-intercom">
<condition field="destination_number" expression="^8(10[01][0-9])$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="sip_auto_answer=true"/>
<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
</condition>
</extension>
<!--
dial the extension (1000-1019) for 30 seconds and go to voicemail if the
call fails (continue_on_fail=true), otherwise hang up after a successful
bridge (hangup_after_bridge=true)
-->
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
<extension name="Local_Extension_Skinny">
<condition field="destination_number" expression="^(11[01][0-9])$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="call_timeout=30"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="skinny/internal/${destination_number}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
<extension name="group_dial_sales">
<condition field="destination_number" expression="^2000$">
<action application="bridge" data="${group_call(sales@${domain_name})}"/>
</condition>
</extension>
<extension name="group_dial_support">
<condition field="destination_number" expression="^2001$">
<action application="bridge" data="group/support@${domain_name}"/>
</condition>
</extension>
<extension name="group_dial_billing">
<condition field="destination_number" expression="^2002$">
<action application="bridge" data="group/billing@${domain_name}"/>
</condition>
</extension>
<!-- voicemail operator extension -->
<extension name="operator">
<condition field="destination_number" expression="^(operator|0)$">
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="transfer" data="1000 XML features"/>
</condition>
</extension>
<!-- voicemail main extension -->
<extension name="vmain">
<condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default ${domain_name}"/>
</condition>
</extension>
<!--
This extension is used by mod_portaudio so you can pa call sip:someone@example.com
mod_portaudio will pass the entire string to the dialplan for routing.
-->
<extension name="sip_uri">
<condition field="destination_number" expression="^sip:(.*)$">
<action application="bridge" data="sofia/${use_profile}/$1"/>
</condition>
</extension>
<!--
start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml
-->
<extension name="nb_conferences">
<condition field="destination_number" expression="^(30\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@default"/>
</condition>
</extension>
<extension name="wb_conferences">
<condition field="destination_number" expression="^(31\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@wideband"/>
</condition>
</extension>
<extension name="uwb_conferences">
<condition field="destination_number" expression="^(32\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@ultrawideband"/>
</condition>
</extension>
<!-- MONO 48kHz conferences -->
<extension name="cdquality_conferences">
<condition field="destination_number" expression="^(33\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@cdquality"/>
</condition>
</extension>
<!-- STEREO 48kHz conferences / Video MCU -->
<extension name="cdquality_stereo_conferences">
<condition field="destination_number" expression="^(35\d{2}).*?-screen$">
<action application="answer"/>
<action application="send_display" data="FreeSWITCH Conference|$1"/>
<action application="set" data="conference_member_flags=join-vid-floor"/>
<action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
</condition>
</extension>
<extension name="conference-canvases" continue="true">
<condition field="destination_number" expression="(35\d{2})-canvas-(\d+)">
<action application="push" data="conference_member_flags=second-screen"/>
<action application="set" data="video_initial_watching_canvas=$2"/>
<action application="transfer" data="$1"/>
</condition>
</extension>
<extension name="conf mod">
<condition field="destination_number" expression="^6070-moderator$">
<action application="answer"/>
<action application="set" data="conference_member_flags=moderator"/>
<action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
</condition>
</extension>
<extension name="cdquality_conferences">
<condition field="destination_number" expression="^(35\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
</condition>
</extension>
<!-- dial the FreeSWITCH conference via SIP-->
<extension name="freeswitch_public_conf_via_sip">
<condition field="destination_number" expression="^9(888|8888|1616|3232)$">
<action application="export" data="hold_music=silence"/>
<!--
This will take the SAS from the b-leg and send it to the display on the a-leg phone.
Known working with Polycom and Snom maybe others.
-->
<!--
<action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/>
<action application="export" data="nolocal:zrtp_secure_media=true"/>
-->
<action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/>
</condition>
</extension>
<!--
This extension will start a conference and invite a group.
At anytime the participant can dial *2 to bridge directly to the boss.
All other callers are then hung up on.
-->
<extension name="mad_boss_intercom">
<condition field="destination_number" expression="^0911$">
<action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/>
<action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
<action application="set" data="conference_auto_outcall_timeout=60"/>
<action application="set" data="conference_auto_outcall_flags=mute"/>
<action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
<action application="set" data="sip_exclude_contact=${network_addr}"/>
<action application="conference_set_auto_outcall" data="${group_call(sales)}"/>
<action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/>
</condition>
</extension>
<!--
This extension will start a conference and invite a few of people.
At anytime the participant can dial *2 to bridge directly to the boss.
All other callers are then hung up on.
-->
<extension name="mad_boss_intercom">
<condition field="destination_number" expression="^0912$">
<action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/>
<action application="set" data="conference_auto_outcall_caller_id_number=0912"/>
<action application="set" data="conference_auto_outcall_timeout=60"/>
<action application="set" data="conference_auto_outcall_flags=mute"/>
<action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
<action application="set" data="sip_exclude_contact=${network_addr}"/>
<action application="conference_set_auto_outcall" data="loopback/9664"/>
<action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/>
</condition>
</extension>
<!--This extension will start a conference and invite several people upon entering -->
<extension name="mad_boss">
<condition field="destination_number" expression="^0913$">
<!--These params effect the outcalls made once you join-->
<action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/>
<action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
<action application="set" data="conference_auto_outcall_timeout=60"/>
<action application="set" data="conference_auto_outcall_flags=none"/>
<!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>-->
<!--Add as many of these as you need, These are the people you are going to call-->
<action application="conference_set_auto_outcall" data="loopback/9664"/>
<action application="conference" data="madboss3@default"/>
</condition>
</extension>
<!-- a sample IVR -->
<extension name="ivr_demo">
<condition field="destination_number" expression="^5000$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
<!-- Create a conference on the fly and pull someone in at the same time. -->
<extension name="dynamic_conference">
<condition field="destination_number" expression="^5001$">
<action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/>
</condition>
</extension>
<extension name="rtp_multicast_page">
<condition field="destination_number" expression="^pagegroup$|^7243$">
<action application="answer"/>
<action application="esf_page_group"/>
</condition>
</extension>
<!--
Parking extensions... transferring calls to 5900 will park them in a queue.
-->
<extension name="park">
<condition field="destination_number" expression="^5900$">
<action application="set" data="fifo_music=$${hold_music}"/>
<action application="fifo" data="5900@${domain_name} in"/>
</condition>
</extension>
<!--
Parking pickup extension. Calling 5901 will pickup the call.
-->
<extension name="unpark">
<condition field="destination_number" expression="^5901$">
<action application="answer"/>
<action application="fifo" data="5900@${domain_name} out nowait"/>
</condition>
</extension>
<!--
Valet park retrieval, works with valet_park extension below.
Retrieve a valet parked call by dialing 6000 + park number + #
-->
<extension name="valet_park">
<condition field="destination_number" expression="^(6000)$">
<action application="answer"/>
<action application="valet_park" data="valet_parking_lot ask 1 11 10000 ivr/ivr-enter_ext_pound.wav"/>
</condition>
</extension>
<!--
Valet park 6001-6099. Blind x-fer to 6001, 6002, etc. to valet park the call.
Dial 6001, 6002, etc. to retrieve a call that is already valet parked.
After call is retrieved, park extension is free for another call.
-->
<extension name="valet_park">
<condition field="destination_number" expression="^((?!6000)60\d{2})$">
<action application="answer"/>
<action application="valet_park" data="valet_parking_lot $1"/>
</condition>
</extension>
<!--
This extension is used with Snom phones.
Set a function key to park+lot (lot being a number or name.)
Set type to Park+Orbit. You can then park and pickup using
the softkey on the phone. Should work with other phones.
-->
<extension name="park">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="park\+(\d+)">
<action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
</condition>
</extension>
<!--
The extension is parking pickup with a to param of the fifo we are calling
Some phones send things like orbit= and you can extract that info.
-->
<extension name="unpark">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^parking$"/>
<condition field="${sip_to_params}" expression="fifo\=(\d+)">
<action application="answer"/>
<action application="fifo" data="$1@${domain_name} out nowait"/>
</condition>
</extension>
<!--
This extension is used with Linksys phones.
Set a Phone tab option Call Park Serv to yes. You can park and
pickup using soft keys "park" and "unpark" found during
active call when moving navigation button. The other option
is to use phone's star codes (defaults to *38 and *39).
-->
<extension name="park">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="callpark"/>
<condition field="${sip_refer_to}">
<expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression>
<action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
</condition>
</extension>
<!--
This extension is used with Linksys phones.
The extension is parking pickup with a to param of the fifo
we are calling. Linksys sends orbit=<parkingslotnumber>
and we extract that info.
-->
<extension name="unpark">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="pickup"/>
<condition field="${sip_to_params}" expression="orbit\=(\d+)">
<action application="answer"/>
<action application="fifo" data="$1@${domain_name} out nowait"/>
</condition>
</extension>
<!--
Here are some examples of how to override the ringback heard by the
far end. You have two variables that you can use to override this.
ringback - used when a call isn't answered. (early media)
transfer_ringback - used when the call is already answered. (post answer)
-->
<!-- Demonstration of how to override the ringback in various situations -->
<extension name="wait">
<condition field="destination_number" expression="^wait$">
<action application="pre_answer"/>
<action application="sleep" data="20000"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="voicemail/vm-goodbye.wav"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="fax_receive">
<condition field="destination_number" expression="^9178$">
<action application="answer" />
<action application="playback" data="silence_stream://2000"/>
<action application="rxfax" data="$${temp_dir}/rxfax.tif"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="fax_transmit">
<condition field="destination_number" expression="^9179$">
<action application="txfax" data="$${temp_dir}/txfax.tif"/>
<action application="hangup"/>
</condition>
</extension>
<!-- Send a 180 and let the far end generate ringback. -->
<extension name="ringback_180">
<condition field="destination_number" expression="^9180$">
<action application="ring_ready"/>
<action application="sleep" data="20000"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="voicemail/vm-goodbye.wav"/>
<action application="hangup"/>
</condition>
</extension>
<!-- Send a 183 and send uk-ring as the ringtone. (early media) -->
<extension name="ringback_183_uk_ring">
<condition field="destination_number" expression="^9181$">
<action application="set" data="ringback=$${uk-ring}"/>
<action application="bridge" data="{ignore_early_media=true}loopback/wait"/>
</condition>
</extension>
<!-- Send a 183 and use music as the ringtone. (early media) -->
<extension name="ringback_183_music_ring">
<condition field="destination_number" expression="^9182$">
<action application="set" data="ringback=$${hold_music}"/>
<action application="bridge" data="{ignore_early_media=true}loopback/wait"/>
</condition>
</extension>
<!-- Answer the call and use music as the ringtone. (post answer) -->
<extension name="ringback_post_answer_uk_ring">
<condition field="destination_number" expression="^9183$">
<action application="set" data="transfer_ringback=$${uk-ring}"/>
<action application="answer"/>
<action application="bridge" data="{ignore_early_media=true}loopback/wait"/>
</condition>
</extension>
<!-- Answer the call and use music as the ringtone. (post answer) -->
<extension name="ringback_post_answer_music">
<condition field="destination_number" expression="^9184$">
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="answer"/>
<action application="bridge" data="{ignore_early_media=true}loopback/wait"/>
</condition>
</extension>
<extension name="ClueCon">
<condition field="destination_number" expression="^9191$">
<action application="set" data="effective_caller_id_name=ClueCon IVR"/>
<action application="bridge" data="sofia/$${domain}/2000@bkw.org"/>
</condition>
</extension>
<extension name="show_info">
<condition field="destination_number" expression="^9192$">
<action application="answer"/>
<action application="info"/>
<action application="sleep" data="250"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="video_record">
<condition field="destination_number" expression="^9193$">
<action application="answer"/>
<action application="record_fsv" data="$${temp_dir}/testrecord.fsv"/>
</condition>
</extension>
<extension name="video_playback">
<condition field="destination_number" expression="^9194$">
<action application="answer"/>
<action application="play_fsv" data="$${temp_dir}/testrecord.fsv"/>
</condition>
</extension>
<extension name="delay_echo">
<condition field="destination_number" expression="^9195$">
<action application="answer"/>
<action application="delay_echo" data="5000"/>
</condition>
</extension>
<extension name="echo">
<condition field="destination_number" expression="^9196$">
<action application="answer"/>
<action application="echo"/>
</condition>
</extension>
<extension name="milliwatt">
<condition field="destination_number" expression="^9197$">
<action application="answer"/>
<action application="playback" data="{loops=-1}tone_stream://%(251,0,1004)"/>
</condition>
</extension>
<extension name="tone_stream">
<condition field="destination_number" expression="^9198$">
<action application="answer"/>
<action application="playback" data="{loops=10}tone_stream://path=${conf_dir}/tetris.ttml"/>
</condition>
</extension>
<!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
<extension name="zrtp_enrollement">
<condition field="destination_number" expression="^9787$">
<action application="lua" data="zrtp_agent.lua"/>
</condition>
</extension>
<!--
You will no longer hear the bong tone. The wav file is playing stating the call is secure.
The file will not play unless you have both TLS and SRTP active.
-->
<extension name="hold_music">
<condition field="destination_number" expression="^9664$"/>
<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
<action application="answer"/>
<action application="execute_extension" data="is_secure XML features"/>
<action application="playback" data="$${hold_music}"/>
<anti-action application="set" data="zrtp_secure_media=true"/>
<anti-action application="answer"/>
<anti-action application="playback" data="silence_stream://2000"/>
<anti-action application="execute_extension" data="is_zrtp_secure XML features"/>
<anti-action application="playback" data="$${hold_music}"/>
</condition>
</extension>
<extension name="laugh break">
<condition field="destination_number" expression="^9386$">
<action application="answer"/>
<action application="sleep" data="1500"/>
<action application="playback" data="phrase:funny_prompts"/>
<action application="hangup"/>
</condition>
</extension>
<!--
You can place files in the default directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
<!--
<extension name="refer">
<condition field="${sip_refer_to}">
<expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression>
</condition>
<condition field="${sip_refer_to}">
<expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
<action application="set" data="refer_user=$1"/>
<action application="set" data="refer_domain=$2"/>
<action application="info"/>
<action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/>
</condition>
</extension>
-->
<!--
This is an example of how to override the RURI on an outgoing invite to a registered contact.
-->
<!--
<extension name="ruri">
<condition field="destination_number" expression="^ruri$">
<action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
</condition>
</extension>
<extension name="7004">
<condition field="destination_number" expression="^7004$">
<action application="set" data="ruri_profile=default"/>
<action application="set" data="ruri_user=2000"/>
<action application="set" data="ruri_contact=1001@${domain_name}"/>
<action application="execute_extension" data="ruri"/>
</condition>
</extension>
-->
<extension name="enum">
<condition field="${module_exists(mod_enum)}" expression="true"/>
<condition field="destination_number" expression="^(.*)$">
<action application="transfer" data="$1 enum"/>
</condition>
</extension>
</context>
</include>

@ -0,0 +1,422 @@
<profile name="internal">
<!--
This is a sofia sip profile/user agent. This will service exactly one ip and port.
In FreeSWITCH you can run multiple sip user agents on their own ip and port.
When you hear someone say "sofia profile" this is what they are talking about.
-->
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!--aliases are other names that will work as a valid profile name for this profile-->
<aliases>
<!--
<alias name="default"/>
-->
</aliases>
<!-- Outbound Registrations -->
<gateways>
</gateways>
<domains>
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
<domain name="$${domain}" parse="true"/>
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
<domain name="all" alias="true" parse="false"/>
</domains>
<settings>
<!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
<!-- <param name="rtp-digit-delay" value="40"/>-->
<!--
When calls are in no media this will bring them back to media
when you press the hold button.
-->
<!--<param name="media-option" value="resume-media-on-hold"/> -->
<!--
This will allow a call after an attended transfer go back to
bypass media after an attended transfer.
-->
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
<!-- Can be set to "_undef_" to remove the User-Agent header -->
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown-on-fail" value="true"/> -->
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
<!-- <param name="presence-proto-lookup" value="true"/> -->
<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
<!--<param name="liberal-dtmf" value="true"/>-->
<!--
Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
responding. These options allow you to enable and control a watchdog
on the Sofia SIP stack so that if it stops responding for the
specified number of milliseconds, it will cause FreeSWITCH to crash
immediately. This is useful if you run in an HA environment and
need to ensure automated recovery from such a condition. Note that if
your server is idle a lot, the watchdog may fire due to not receiving
any SIP messages. Thus, if you expect your system to be idle, you
should leave the watchdog disabled. It can be toggled on and off
through the FreeSWITCH CLI either on an individual profile basis or
globally for all profiles. So, if you run in an HA environment with a
master and slave, you should use the CLI to make sure the watchdog is
only enabled on the master.
If such crash occurs, FreeSWITCH will dump core if allowed. The
stacktrace will include function watchdog_triggered_abort().
-->
<param name="watchdog-enabled" value="no"/>
<param name="watchdog-step-timeout" value="30000"/>
<param name="watchdog-event-timeout" value="30000"/>
<param name="log-auth-failures" value="false"/>
<param name="forward-unsolicited-mwi-notify" value="false"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="$${internal_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="rtp-timer-name" value="soft"/>
<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="apply-nat-acl" value="nat.auto"/>
<!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
<!-- <param name="cid-in-1xx" value="false"/> -->
<!-- extended info parsing -->
<!-- <param name="extended-info-parsing" value="true"/> -->
<!--<param name="aggressive-nat-detection" value="true"/>-->
<!--
There are known issues (asserts and segfaults) when 100rel is enabled.
It is not recommended to enable 100rel at this time.
-->
<!--<param name="enable-100rel" value="true"/>-->
<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
<!-- RFC3263 Section 4.3 -->
<!--<param name="disable-srv503" value="true"/>-->
<!-- Enable Compact SIP headers. -->
<!--<param name="enable-compact-headers" value="true"/>-->
<!--
enable/disable session timers
-->
<!--<param name="enable-timer" value="false"/>-->
<!--<param name="minimum-session-expires" value="120"/>-->
<!-- <param name="apply-inbound-acl" value="domains"/>-->
<!--
This defines your local network, by default we detect your local network
and create this localnet.auto ACL for this.
-->
<param name="local-network-acl" value="localnet.auto"/>
<!--<param name="apply-register-acl" value="domains"/>-->
<!--<param name="dtmf-type" value="info"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-message-query-on-register" value="true"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-presence-on-register" value="first-only"/> -->
<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
<!-- Remote-Party-ID header -->
<!--<param name="caller-id-type" value="rpid"/>-->
<!-- P-*-Identity family of headers -->
<!--<param name="caller-id-type" value="pid"/>-->
<!-- neither one -->
<!--<param name="caller-id-type" value="none"/>-->
<param name="record-path" value="$${recordings_dir}"/>
<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<!--enable to use presence -->
<param name="manage-presence" value="true"/>
<!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
<!--<param name="presence-probe-on-register" value="true"/>-->
<!--<param name="manage-shared-appearance" value="true"/>-->
<!-- used to share presence info across sofia profiles -->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
<param name="presence-privacy" value="$${presence_privacy}"/>
<!-- ************************************************* -->
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<!--<param name="max-proceeding" value="1000"/>-->
<!--session timers for all call to expire after the specified seconds -->
<!--<param name="session-timeout" value="1800"/>-->
<!-- Can be 'true' or 'contact' -->
<!--<param name="multiple-registrations" value="contact"/>-->
<!--set to 'greedy' if you want your codec list to take precedence -->
<param name="inbound-codec-negotiation" value="generous"/>
<!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--<param name="unregister-on-options-fail" value="true"/>-->
<!-- Send an OPTIONS packet to all registered endpoints -->
<!--<param name="all-reg-options-ping" value="true"/>-->
<!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
<!--<param name="nat-options-ping" value="true"/>-->
<!--<param name="sip-options-respond-503-on-busy" value="true"/>-->
<!--<param name="sip-messages-respond-200-ok" value="true"/>-->
<!--<param name="sip-subscribe-respond-200-ok" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${internal_ssl_enable}"/>
<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
<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="$${internal_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<!--<param name="tls-cert-dir" value=""/>-->
<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
<param name="tls-passphrase" value=""/>
<!-- Verify the date on TLS certificates -->
<param name="tls-verify-date" value="true"/>
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'subjects_in', 'subjects_out' and 'subjects_all' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="none"/>
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name="tls-verify-depth" value="2"/>
<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
<param name="tls-verify-in-subjects" value=""/>
<!-- TLS version default: tlsv1,tlsv1.1,tlsv1.2 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH -->
<param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
(reduces delay on latent connections default true, must be disabled explicitly)-->
<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
<!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!-- Or, if you have PGSQL support, you can use that -->
<!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-media" value="true"/>-->
<!-- Let calls hit the dialplan before selecting codec for the a-leg -->
<param name="inbound-late-negotiation" value="true"/>
<!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
<param name="inbound-zrtp-passthru" value="true"/>
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
<!-- <param name="accept-blind-reg" value="true"/> -->
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<!-- <param name="accept-blind-auth" value="true"/> -->
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
<!-- <param name="suppress-cng" value="true"/> -->
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!-- Handle 302 Redirect in the dialplan -->
<!--<param name="manual-redirect" value="true"/> -->
<!-- Disable Transfer -->
<!--<param name="disable-transfer" value="true"/> -->
<!-- Disable Register -->
<!--<param name="disable-register" value="true"/> -->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
<param name="auth-calls" value="$${internal_auth_calls}"/>
<!-- Force the user and auth-user to match. -->
<param name="inbound-reg-force-matching-username" value="true"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an one of:
ip address - "12.34.56.78"
a stun server lookup - "stun:stun.server.com"
a DNS name - "host:host.server.com"
auto - Use guessed ip.
auto-nat - Use ip learned from NAT-PMP or UPNP
-->
<param name="ext-rtp-ip" value="127.0.0.1"/>
<param name="ext-sip-ip" value="127.0.0.1"/>
<!-- rtp inactivity timeout -->
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
<!--
These are enabled to make the default config work better out of the box.
If you need more than ONE domain you'll need to not use these options.
-->
<!--all inbound reg will look in this domain for the users -->
<param name="force-register-domain" value="$${domain}"/>
<!--force the domain in subscriptions to this value -->
<param name="force-subscription-domain" value="$${domain}"/>
<!--all inbound reg will stored in the db using this domain -->
<param name="force-register-db-domain" value="$${domain}"/>
<!-- for sip over websocket support -->
<param name="ws-binding" value=":5066"/>
<!-- for sip over secure websocket support -->
<!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
<param name="wss-binding" value=":7443"/>
<!--<param name="delete-subs-on-register" value="false"/>-->
<!-- launch a new thread to process each new inbound register when using heavier backends -->
<!-- <param name="inbound-reg-in-new-thread" value="true"/> -->
<!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
<!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
<!--<param name="rtcp-video-interval-msec" value="5000"/>-->
<!--force suscription expires to a lower value than requested-->
<!--<param name="force-subscription-expires" value="60"/>-->
<!-- add a random deviation to the expires value of the 202 Accepted -->
<!--<param name="sip-subscription-max-deviation" value="120"/>-->
<!-- disable register and transfer which may be undesirable in a public switch -->
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!--
enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
right away, proxy waits until the call has been answered then sends accepts
-->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use at your own risk or if you know what this does.-->
<!--<param name="NDLB-force-rport" value="true"/>-->
<!--
Choose the realm challenge key. Default is auto_to if not set.
auto_from - uses the from field as the value for the sip realm.
auto_to - uses the to field as the value for the sip realm.
<anyvalue> - you can input any value to use for the sip realm.
If you want URL dialing to work you'll want to set this to auto_from.
If you use any other value besides auto_to or auto_from you'll
loose the ability to do multiple domains.
Note: comment out to restore the behavior before 2008-09-29
-->
<param name="challenge-realm" value="auto_from"/>
<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
<!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
<!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
<!-- on outbound calls set the callid to match the uuid of the session -->
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
<!-- set to false disable this feature -->
<!--<param name="rtp-autofix-timing" value="false"/>-->
<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
<!--<param name="pass-callee-id" value="false"/>-->
<!-- clear clears them all or supply the name to add or the name
prefixed with ~ to remove valid values:
clear
CISCO_SKIP_MARK_BIT_2833
SONUS_SEND_INVALID_TIMESTAMP_2833
-->
<!--<param name="auto-rtp-bugs" data="clear"/>-->
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
<!--<param name="disable-srv" value="false" />-->
<!--<param name="disable-naptr" value="false" />-->
<!-- The following can be used to fine-tune timers within sofia's transport layer
Those settings are for advanced users and can safely be left as-is -->
<!-- Initial retransmission interval (in milliseconds).
Set the T1 retransmission interval used by the SIP transaction engine.
The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
<!-- <param name="timer-T1" value="500" /> -->
<!-- Transaction timeout (defaults to T1 * 64).
Set the T1x64 timeout value used by the SIP transaction engine.
The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
<!-- <param name="timer-T1X64" value="32000" /> -->
<!-- Maximum retransmission interval (in milliseconds).
Set the maximum retransmission interval used by the SIP transaction engine.
The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
until the timer B fires. -->
<!-- <param name="timer-T2" value="4000" /> -->
<!--
Transaction lifetime (in milliseconds).
Set the lifetime for completed transactions used by the SIP transaction engine.
A completed transaction is kept around for the duration of T4 in order to catch late responses.
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
<!-- <param name="timer-T4" value="4000" /> -->
<!-- Turn on a jitterbuffer for every call -->
<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
It's probably not what you want so stick with the default unless you really need to change this.
-->
<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
</settings>
</profile>

@ -0,0 +1,68 @@
<!--
NOTICE:
This context is usually accessed via the external sip profile listening on port 5080.
It is recommended to have separate inbound and outbound contexts. Not only for security
but clearing up why you would need to do such a thing. You don't want outside un-authenticated
callers hitting your default context which allows dialing calls thru your providers and results
in Toll Fraud.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!--
Tag anything pass thru here as an outside_call so you can make sure not
to create any routing loops based on the conditions that it came from
the outside of the switch.
-->
<extension name="outside_call" continue="true">
<condition>
<action application="set" data="outside_call=true"/>
<action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
</condition>
</extension>
<extension name="call_debug" continue="true">
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
</extension>
<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<!--
You can place files in the public directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="public/*.xml"/>
<!--
If you have made it this far lets challenge the caller and if they authenticate
lets try what they dialed in the default context. (commented out by default)
-->
<!--
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>
-->
<extension name="transfer_to_default">
<condition>
<action application="transfer" data="${destination_number} XML default"/>
</condition>
</extension>
</context>
</include>

@ -0,0 +1,181 @@
<configuration name="switch.conf" description="Core Configuration">
<cli-keybindings>
<key name="1" value="help"/>
<key name="2" value="status"/>
<key name="3" value="show channels"/>
<key name="4" value="show calls"/>
<key name="5" value="sofia status"/>
<key name="6" value="reloadxml"/>
<key name="7" value="console loglevel 0"/>
<key name="8" value="console loglevel 7"/>
<key name="9" value="sofia status profile internal"/>
<key name="10" value="sofia profile internal siptrace on"/>
<key name="11" value="sofia profile internal siptrace off"/>
<key name="12" value="version"/>
</cli-keybindings>
<default-ptimes>
<!-- Set this to override the 20ms assumption of various codecs in the sdp with no ptime defined -->
<!-- <codec name="G729" ptime="40"/> -->
</default-ptimes>
<settings>
<!-- Colorize the Console -->
<param name="colorize-console" value="true"/>
<!--Include full timestamps in dialplan logs -->
<param name="dialplan-timestamps" value="false"/>
<!-- Run the timer at 20ms by default and drop down as needed unless you set 1m-timer=true which was previous default -->
<!-- <param name="1ms-timer" value="true"/> -->
<!--
Set the Switch Name for HA environments.
When setting the switch name, it will override the system hostname for all DB and CURL requests
allowing cluster environments such as RHCS to have identical FreeSWITCH configurations but run
as different hostnames.
-->
<!-- <param name="switchname" value="freeswitch"/> -->
<!-- <param name="cpu-idle-smoothing-depth" value="30"/> -->
<!-- Maximum number of simultaneous DB handles open -->
<param name="max-db-handles" value="50"/>
<!-- Maximum number of seconds to wait for a new DB handle before failing -->
<param name="db-handle-timeout" value="10"/>