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authorHolger Hans Peter Freyther <holger@moiji-mobile.com>2017-03-05 16:47:57 +0100
committerHolger Hans Peter Freyther <holger@moiji-mobile.com>2017-03-05 17:01:21 +0100
commit7166d0f448eae451b1a6287b522c0fc82387b05b (patch)
tree57c749e1cf281fb74d03001f32d71c2253f0f129 /contrib
parent42b073a233740e0e0125e99e4bea29ac7d3d27ed (diff)
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
Diffstat (limited to 'contrib')
-rw-r--r--contrib/testpbx/Dockerfile25
-rw-r--r--contrib/testpbx/Makefile12
-rw-r--r--contrib/testpbx/README29
-rw-r--r--contrib/testpbx/configs/acl.conf.xml34
-rw-r--r--contrib/testpbx/configs/default.xml832
-rw-r--r--contrib/testpbx/configs/internal.xml422
-rw-r--r--contrib/testpbx/configs/public.xml68
-rw-r--r--contrib/testpbx/configs/switch.conf.xml181
-rw-r--r--contrib/testpbx/configs/vars.xml450
9 files changed, 2053 insertions, 0 deletions
diff --git a/contrib/testpbx/Dockerfile b/contrib/testpbx/Dockerfile
new file mode 100644
index 0000000..2f03424
--- /dev/null
+++ b/contrib/testpbx/Dockerfile
@@ -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
diff --git a/contrib/testpbx/Makefile b/contrib/testpbx/Makefile
new file mode 100644
index 0000000..ea34799
--- /dev/null
+++ b/contrib/testpbx/Makefile
@@ -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
diff --git a/contrib/testpbx/README b/contrib/testpbx/README
new file mode 100644
index 0000000..11c16f0
--- /dev/null
+++ b/contrib/testpbx/README
@@ -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
diff --git a/contrib/testpbx/configs/acl.conf.xml b/contrib/testpbx/configs/acl.conf.xml
new file mode 100644
index 0000000..70a64ea
--- /dev/null
+++ b/contrib/testpbx/configs/acl.conf.xml
@@ -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>
diff --git a/contrib/testpbx/configs/default.xml b/contrib/testpbx/configs/default.xml
new file mode 100644
index 0000000..f0e0af1
--- /dev/null
+++ b/contrib/testpbx/configs/default.xml
@@ -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>
diff --git a/contrib/testpbx/configs/internal.xml b/contrib/testpbx/configs/internal.xml
new file mode 100644
index 0000000..2a679fb
--- /dev/null
+++ b/contrib/testpbx/configs/internal.xml
@@ -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>
diff --git a/contrib/testpbx/configs/public.xml b/contrib/testpbx/configs/public.xml
new file mode 100644
index 0000000..d9b1d17
--- /dev/null
+++ b/contrib/testpbx/configs/public.xml
@@ -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>
diff --git a/contrib/testpbx/configs/switch.conf.xml b/contrib/testpbx/configs/switch.conf.xml
new file mode 100644
index 0000000..1a82409
--- /dev/null
+++ b/contrib/testpbx/configs/switch.conf.xml
@@ -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"/>
+
+ <!-- Minimum idle CPU before refusing calls -->
+ <!-- <param name="min-idle-cpu" value="25"/> -->
+
+ <!--
+ Max number of sessions to allow at any given time.
+
+ NOTICE: If you're driving 28 T1's in a single box you should set this to 644*2 or 1288
+ this will ensure you're able to use the entire DS3 without a problem. Otherwise you'll
+ be 144 channels short of always filling that DS3 up which can translate into waste.
+ -->
+ <param name="max-sessions" value="1000"/>
+ <!--Most channels to create per second -->
+ <param name="sessions-per-second" value="30"/>
+ <!-- Default Global Log Level - value is one of debug,info,notice,warning,err,crit,alert -->
+ <param name="loglevel" value="debug"/>
+
+ <!-- Set the core DEBUG level (0-10) -->
+ <!-- <param name="debug-level" value="10"/> -->
+
+ <!-- SQL Buffer length within rage of 32k to 10m -->
+ <!-- <param name="sql-buffer-len" value="1m"/> -->
+ <!-- Maximum SQL Buffer length must be greater than sql-buffer-len -->
+ <!-- <param name="max-sql-buffer-len" value="2m"/> -->
+
+ <!--
+ The min-dtmf-duration specifies the minimum DTMF duration to use on
+ outgoing events. Events shorter than this will be increased in duration
+ to match min_dtmf_duration. You cannot configure a dtmf duration on a
+ profile that is less than this setting. You may increase this value,
+ but cannot set it lower than 400. This value cannot exceed
+ max-dtmf-duration. -->
+ <!-- <param name="min-dtmf-duration" value="400"/> -->
+
+ <!--
+ The max-dtmf-duration caps the playout of a DTMF event at the specified
+ duration. Events exceeding this duration will be truncated to this
+ duration. You cannot configure a duration on a profile that exceeds
+ this setting. This setting can be lowered, but cannot exceed 192000.
+ This setting cannot be set lower than min_dtmf_duration. -->
+ <!-- <param name="max-dtmf-duration" value="192000"/> -->
+
+ <!--
+ The default_dtmf_duration specifies the DTMF duration to use on
+ originated DTMF events or on events that are received without a
+ duration specified. This value can be increased or lowered. This
+ value is lower-bounded by min_dtmf_duration and upper-bounded by
+ max-dtmf-duration\. -->
+ <!-- <param name="default-dtmf-duration" value="2000"/> -->
+
+ <!--
+ If you want to send out voicemail notifications via Windows you'll need to change the mailer-app
+ variable to the setting below:
+
+ <param name="mailer-app" value="msmtp"/>
+
+ Do not change mailer-app-args.
+ You will also need to download a sendmail clone for Windows (msmtp). This version works without issue:
+ http://msmtp.sourceforge.net/index.html. Download and copy the .exe to %winddir%\system32.
+ You'll need to create a small config file for smtp credentials (host name, authentication, tls, etc.) in
+ %USERPROFILE%\Application Data\ called "msmtprc.txt". Below is a sample copy of this file:
+
+ ###################################
+ # The SMTP server of the provider.
+ account provider
+ host smtp.myisp.com
+ from john@myisp.com
+ auth login
+ user johndoe
+ password mypassword
+
+ # Set a default account
+ account default : provider
+ ###################################
+
+ -->
+
+ <param name="mailer-app" value="sendmail"/>
+ <param name="mailer-app-args" value="-t"/>
+ <param name="dump-cores" value="yes"/>
+
+ <!-- Enable verbose channel events to include every detail about a channel on every event -->
+ <!-- <param name="verbose-channel-events" value="no"/> -->
+
+ <!-- Enable clock nanosleep -->
+ <!-- <param name="enable-clock-nanosleep" value="true"/> -->
+
+ <!-- Enable monotonic timing -->
+ <!-- <param name="enable-monotonic-timing" value="true"/> -->
+
+ <!-- NEEDS DOCUMENTATION -->
+ <!-- <param name="enable-softtimer-timerfd" value="true"/> -->
+ <!-- <param name="enable-cond-yield" value="true"/> -->
+ <!-- <param name="enable-timer-matrix" value="true"/> -->
+ <!-- <param name="threaded-system-exec" value="true"/> -->
+ <!-- <param name="tipping-point" value="0"/> -->
+ <!-- <param name="timer-affinity" value="disabled"/> -->
+ <!-- NEEDS DOCUMENTATION -->
+
+ <!-- RTP port range -->
+ <param name="rtp-start-port" value="6000"/>
+ <param name="rtp-end-port" value="6020"/>
+
+ <!-- Test each port to make sure it is not in use by some other process before allocating it to RTP -->
+ <!-- <param name="rtp-port-usage-robustness" value="true"/> -->
+
+ <param name="rtp-enable-zrtp" value="false"/>
+
+ <!--
+ Store encryption keys for secure media in channel variables and call CDRs. Default: false.
+ WARNING: If true, anyone with CDR access can decrypt secure media!
+ -->
+ <!-- <param name="rtp-retain-crypto-keys" value="true"/> -->
+
+ <!-- <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE'" /> -->
+ <!-- <param name="core-db-dsn" value="dsn:username:password" /> -->
+ <!--
+ Allow to specify the sqlite db at a different location (In this example, move it to ramdrive for
+ better performance on most linux distro (note, you loose the data if you reboot))
+ -->
+ <!-- <param name="core-db-name" value="/dev/shm/core.db" /> -->
+
+ <!-- The system will create all the db schemas automatically, set this to false to avoid this behaviour -->
+ <!-- <param name="auto-create-schemas" value="true"/> -->
+ <!-- <param name="auto-clear-sql" value="true"/> -->
+ <!-- <param name="enable-early-hangup" value="true"/> -->
+
+ <!-- <param name="core-dbtype" value="MSSQL"/> -->
+
+ <!-- Allow multiple registrations to the same account in the central registration table -->
+ <!-- <param name="multiple-registrations" value="true"/> -->
+
+ </settings>
+
+</configuration>
diff --git a/contrib/testpbx/configs/vars.xml b/contrib/testpbx/configs/vars.xml
new file mode 100644
index 0000000..1cb826d
--- /dev/null
+++ b/contrib/testpbx/configs/vars.xml
@@ -0,0 +1,450 @@
+<include>
+ <!-- Preprocessor Variables
+ These are introduced when configuration strings must be consistent across modules.
+ NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+
+ YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
+ toll fraud in the future. It's your responsibility to secure your own system.
+
+ This default config is used to demonstrate the feature set of FreeSWITCH.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+ -->
+ <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
+ <!-- Did you change it yet? -->
+ <!--
+ The following variables are set dynamically - calculated if possible by freeswitch - and
+ are available to the config as $${variable}. You can see their calculated value via fs_cli
+ by entering eval $${variable}
+
+ hostname
+ local_ip_v4
+ local_mask_v4
+ local_ip_v6
+ switch_serial
+ base_dir
+ recordings_dir
+ sound_prefix
+ sounds_dir
+ conf_dir
+ log_dir
+ run_dir
+ db_dir
+ mod_dir
+ htdocs_dir
+ script_dir
+ temp_dir
+ grammar_dir
+ certs_dir
+ storage_dir
+ cache_dir
+ core_uuid
+ zrtp_enabled
+ nat_public_addr
+ nat_private_addr
+ nat_type
+
+ -->
+
+
+ <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
+
+ <!--
+ This setting is what sets the default domain FreeSWITCH will use if all else fails.
+
+ FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does
+ affect the sip authentication. Please review conf/directory/default.xml for more
+ information on this topic.
+ -->
+ <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
+ <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
+ <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
+ <X-PRE-PROCESS cmd="set" data="use_profile=external"/>
+ <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
+ <!--
+ Enable ZRTP globally you can override this on a per channel basis
+
+ http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
+ -->
+ <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
+ <!--
+ NOTICE: When using SRTP it's critical that you do not offer or accept
+ variable bit rate codecs, doing so would leak information and possibly
+ compromise your SRTP stream. (FS-6404)
+
+ Supported SRTP Crypto Suites:
+
+ AEAD_AES_256_GCM_8
+ ____________________________________________________________________________
+ This algorithm is identical to AEAD_AES_256_GCM (see Section 5.2 of
+ [RFC5116]), except that the tag length, t, is 8, and an
+ authentication tag with a length of 8 octets (64 bits) is used.
+ An AEAD_AES_256_GCM_8 ciphertext is exactly 8 octets longer than its
+ corresponding plaintext.
+
+
+ AEAD_AES_128_GCM_8
+ ____________________________________________________________________________
+ This algorithm is identical to AEAD_AES_128_GCM (see Section 5.1 of
+ [RFC5116]), except that the tag length, t, is 8, and an
+ authentication tag with a length of 8 octets (64 bits) is used.
+ An AEAD_AES_128_GCM_8 ciphertext is exactly 8 octets longer than its
+ corresponding plaintext.
+
+
+ AES_CM_256_HMAC_SHA1_80 | AES_CM_192_HMAC_SHA1_80 | AES_CM_128_HMAC_SHA1_80
+ ____________________________________________________________________________
+ AES_CM_128_HMAC_SHA1_80 is the SRTP default AES Counter Mode cipher
+ and HMAC-SHA1 message authentication with an 80-bit authentication
+ tag. The master-key length is 128 bits and has a default lifetime of
+ a maximum of 2^48 SRTP packets or 2^31 SRTCP packets, whichever comes
+ first.
+
+
+ AES_CM_256_HMAC_SHA1_32 | AES_CM_192_HMAC_SHA1_32 | AES_CM_128_HMAC_SHA1_32
+ ____________________________________________________________________________
+ This crypto-suite is identical to AES_CM_128_HMAC_SHA1_80 except that
+ the authentication tag is 32 bits. The length of the base64-decoded key and
+ salt value for this crypto-suite MUST be 30 octets i.e., 240 bits; otherwise,
+ the crypto attribute is considered invalid.
+
+
+ AES_CM_128_NULL_AUTH
+ ____________________________________________________________________________
+ The SRTP default cipher (AES-128 Counter Mode), but to use no authentication
+ method. This policy is NOT RECOMMENDED unless it is unavoidable; see
+ Section 7.5 of [RFC3711].
+
+
+ SRTP variables that modify behaviors based on direction/leg:
+
+ rtp_secure_media
+ ____________________________________________________________________________
+ possible values:
+ mandatory - Accept/Offer SAVP negotiation ONLY
+ optional - Accept/Offer SAVP/AVP with SAVP preferred
+ forbidden - More useful for inbound to deny SAVP negotiation
+ false - implies forbidden
+ true - implies mandatory
+
+ default if not set is accept SAVP inbound if offered.
+
+
+ rtp_secure_media_inbound | rtp_secure_media_outbound
+ ____________________________________________________________________________
+ This is the same as rtp_secure_media, but would apply to either inbound
+ or outbound offers specifically.
+
+
+ How to specify crypto suites:
+ ____________________________________________________________________________
+ By default without specifying any crypto suites FreeSWITCH will offer
+ crypto suites from strongest to weakest accepting the strongest each
+ endpoint has in common. If you wish to force specific crypto suites you
+ can do so by appending the suites in a comma separated list in the order
+ that you wish to offer them in.
+
+ Examples:
+
+ rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
+ rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
+ rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
+ rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80
+
+ Additionally you can narrow this down on either inbound or outbound by
+ specifying as so:
+
+ rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8
+ rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8
+ rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8
+ rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8
+
+
+ rtp_secure_media_suites
+ ____________________________________________________________________________
+ Optionaly you can use rtp_secure_media_suites to dictate the suite list
+ and only use rtp_secure_media=[optional|mandatory|false|true] without having
+ to dictate the suite list with the rtp_secure_media* variables.
+ -->
+ <!--
+ Examples of codec options: (module must be compiled and loaded)
+
+ codecname[@8000h|16000h|32000h[@XXi]]
+
+ XX is the frame size must be multples allowed for the codec
+ FreeSWITCH can support 10-120ms on some codecs.
+ We do not support exceeding the MTU of the RTP packet.
+
+
+ iLBC@30i - iLBC using mode=30 which will win in all cases.
+ DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
+ DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
+ speex@8000h@20i - Speex 8kHz using 20ms ptime.
+ speex@16000h@20i - Speex 16kHz using 20ms ptime.
+ speex@32000h@20i - Speex 32kHz using 20ms ptime.
+ BV16 - BroadVoice 16kb/s narrowband, 8kHz
+ BV32 - BroadVoice 32kb/s wideband, 16kHz
+ G7221@16000h - G722.1 16kHz (aka Siren 7)
+ G7221@32000h - G722.1C 32kHz (aka Siren 14)
+ CELT@32000h - CELT 32kHz, only 10ms supported
+ CELT@48000h - CELT 48kHz, only 10ms supported
+ GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
+ G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
+ PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
+ PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
+ G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
+ AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
+ LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
+ L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
+
+ These are the passthru audio codecs:
+
+ G729 - G729 in passthru mode. (mod_g729)
+ G723 - G723.1 in passthru mode. (mod_g723_1)
+ AMR - AMR in passthru mode. (mod_amr)
+
+ These are the passthru video codecs: (mod_h26x)
+
+ H261 - H.261 Video
+ H263 - H.263 Video
+ H263-1998 - H.263-1998 Video
+ H263-2000 - H.263-2000 Video
+ H264 - H.264 Video
+
+ RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.
+
+ 96 - AMR
+ 97 - iLBC (30)
+ 98 - iLBC (20)
+ 99 - Speex 8kHz, 16kHz, 32kHz
+ 100 -
+ 101 - telephone-event
+ 102 -
+ 103 -
+ 104 -
+ 105 -
+ 106 - BV16
+ 107 - G722.1 (16kHz)
+ 108 -
+ 109 -
+ 110 -
+ 111 -
+ 112 -
+ 113 -
+ 114 - CELT 32kHz, 48kHz
+ 115 - G722.1C (32kHz)
+ 116 -
+ 117 - SILK 8kHz
+ 118 - SILK 12kHz
+ 119 - SILK 16kHz
+ 120 - SILK 24kHz
+ 121 - AAL2-G726-40 && G726-40
+ 122 - AAL2-G726-32 && G726-32
+ 123 - AAL2-G726-24 && G726-24
+ 124 - AAL2-G726-16 && G726-16
+ 125 -
+ 126 -
+ 127 - BV32
+
+ -->
+ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
+
+ <!--
+ xmpp_client_profile and xmpp_server_profile
+ xmpp_client_profile can be any string.
+ xmpp_server_profile is appended to "dingaling_" to form the database name
+ containing the "subscriptions" table.
+ used by: dingaling.conf.xml enum.conf.xml
+ -->
+
+ <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
+ <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
+ <!--
+ THIS IS ONLY USED FOR DINGALING
+
+ bind_server_ip
+
+ Can be an ip address, a dns name, or "auto".
+ This determines an ip address available on this host to bind.
+ If you are separating RTP and SIP traffic, you will want to have
+ use different addresses where this variable appears.
+ Used by: dingaling.conf.xml
+ -->
+ <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
+
+ <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
+
+ If you're going to load test FreeSWITCH please input real IP addresses
+ for external_rtp_ip and external_sip_ip
+ -->
+
+ <!-- external_rtp_ip
+ 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"
+ where fs.mydomain.com is a DNS A record-useful when fs is on
+ a dynamic IP address, and uses a dynamic DNS updater.
+ If unspecified, the bind_server_ip value is used.
+ Used by: sofia.conf.xml dingaling.conf.xml
+ -->
+ <!-- <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> -->
+ <X-PRE-PROCESS cmd="set" data="external_rtp_ip=127.0.0.1"/>
+
+ <!-- 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"
+ where fs.mydomain.com is a DNS A record-useful when fs is on
+ a dynamic IP address, and uses a dynamic DNS updater.
+ If unspecified, the bind_server_ip value is used.
+ Used by: sofia.conf.xml dingaling.conf.xml
+ -->
+ <!-- <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> -->
+ <X-PRE-PROCESS cmd="set" data="external_sip_ip=127.0.0.1"/>
+
+
+ <!-- unroll-loops
+ Used to turn on sip loopback unrolling.
+ -->
+ <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
+
+ <!-- outbound_caller_id and outbound_caller_name
+ The caller ID telephone number we should use when calling out.
+ Used by: conference.conf.xml and user directory for default
+ outbound callerid name and number.
+ -->
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
+
+ <!-- various debug and defaults -->
+ <X-PRE-PROCESS cmd="set" data="call_debug=false"/>
+ <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
+ <X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
+ <X-PRE-PROCESS cmd="set" data="default_country=US"/>
+
+ <!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
+ if true, the destination number is not included -->
+ <X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>
+
+ <X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/>
+ <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
+ <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
+ <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
+ <X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
+ <X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
+ <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
+ <X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
+ <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
+ <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
+ <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
+ <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
+ <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
+ <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
+ <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
+ <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
+ <X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/>
+ <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
+
+ <!--
+ Digits Dialed filter: (FS-6940)
+
+ The digits stream may contain valid credit card numbers or social security numbers, These digit
+ filters will allow you to make a valant effort to stamp out sensitive information for
+ PCI/HIPPA compliance. (see xml_cdr dialed_digits)
+
+ df_us_ssn = US Social Security Number pattern
+ df_us_luhn = Visa, MasterCard, American Express, Diners Club, Discover and JCB
+ -->
+ <X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/>
+ <X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/>
+ <!-- change XX to X below to enable -->
+ <XX-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/>
+
+ <!--
+ Setting up your default sip provider is easy.
+ Below are some values that should work in most cases.
+
+ These are for conf/directory/default/example.com.xml
+ -->
+ <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
+ <!-- true or false -->
+ <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
+
+ <!--
+ SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
+
+ 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"/>
+
+ <!--
+ TLS cipher suite: default ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH
+
+ The actual ciphers supported will change per platform.
+
+ openssl ciphers -v 'ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH'
+
+ Will show you what is available in your verion of openssl.
+ -->
+ <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>
+
+ <!-- Internal SIP Profile -->
+ <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
+ <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
+ <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
+ <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
+
+ <!-- External SIP Profile -->
+ <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
+ <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
+ <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
+ <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
+
+ <!-- Video Settings -->
+ <!-- Setting the max bandwdith -->
+ <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=1mb"/>
+ <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=1mb"/>
+
+ <!-- WebRTC Video -->
+ <!-- Suppress CNG for WebRTC Audio -->
+ <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
+ <!-- Enable liberal DTMF for those that can't get it right -->
+ <X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
+ <!-- Helps with WebRTC Audio -->
+
+ <!-- Stock Video Avatars -->
+ <X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/>
+ <X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/>
+
+</include>