diff options
Diffstat (limited to 'contrib')
| -rw-r--r-- | contrib/testpbx/Dockerfile | 25 | ||||
| -rw-r--r-- | contrib/testpbx/Makefile | 12 | ||||
| -rw-r--r-- | contrib/testpbx/README | 29 | ||||
| -rw-r--r-- | contrib/testpbx/configs/acl.conf.xml | 34 | ||||
| -rw-r--r-- | contrib/testpbx/configs/default.xml | 832 | ||||
| -rw-r--r-- | contrib/testpbx/configs/internal.xml | 422 | ||||
| -rw-r--r-- | contrib/testpbx/configs/public.xml | 68 | ||||
| -rw-r--r-- | contrib/testpbx/configs/switch.conf.xml | 181 | ||||
| -rw-r--r-- | contrib/testpbx/configs/vars.xml | 450 |
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> |