[Freeswitch-branches] [commit] r10236 - freeswitch/branches/gmaruzz/stuff
Freeswitch SVN
gmaruzz at freeswitch.org
Tue Nov 4 10:11:45 EST 2008
Author: gmaruzz
Date: Tue Nov 4 10:11:45 2008
New Revision: 10236
Added:
freeswitch/branches/gmaruzz/stuff/
freeswitch/branches/gmaruzz/stuff/default.xml
freeswitch/branches/gmaruzz/stuff/modules.conf.xml
freeswitch/branches/gmaruzz/stuff/openzap.conf.xml
freeswitch/branches/gmaruzz/stuff/portaudio.conf.xml
freeswitch/branches/gmaruzz/stuff/skypiax.conf.xml
Log:
added stuff/configs
Added: freeswitch/branches/gmaruzz/stuff/default.xml
==============================================================================
--- (empty file)
+++ freeswitch/branches/gmaruzz/stuff/default.xml Tue Nov 4 10:11:45 2008
@@ -0,0 +1,603 @@
+<!--
+ 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.
+-->
+
+<?xml version="1.0" encoding="utf-8"?>
+<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
+<include>
+ <context name="default">
+
+ <extension name="unloop">
+ <condition field="$${unroll_loops}" expression="^true$"/>
+ <condition field="${sip_looped_call}" expression="^true$">
+ <action application="deflect" data="${destination_number}"/>
+ </condition>
+ </extension>
+
+ <!--
+ Try to get the domain from the sip_auth_realm otherwise it will
+ default domain in vars.xml for cases it can't figure it out.
+
+ -->
+ <extension name="set_domain" continue="true">
+ <condition field="${domain_name}" expression="^$" break="never"/>
+ <condition field="source" expression="mod_sofia" break="never"/>
+ <condition field="${sip_auth_realm}" expression="^$" break="never">
+ <action application="set" data="domain_name=$${domain}"/>
+ <anti-action application="set" data="domain_name=${sip_auth_realm}"/>
+ </condition>
+ </extension>
+
+ <!-- Example of doing things based on time of day. -->
+ <extension name="tod_example" continue="true">
+ <!-- man strftime - M-F, 9AM to 6PM -->
+ <condition field="${strftime(%w)}" expression="^([1-5])$"/>
+ <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$">
+ <action application="set" data="open=true"/>
+ </condition>
+ </extension>
+
+ <extension name="global-intercept">
+ <condition field="destination_number" expression="^886$">
+ <action application="answer"/>
+ <action application="intercept" data="${db(select/${domain_name}-last_dial/global)}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="group-intercept">
+ <condition field="destination_number" expression="^\*8$">
+ <action application="answer"/>
+ <action application="intercept" data="${db(select/${domain_name}-last_dial/${callgroup})}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="intercept-ext">
+ <condition field="destination_number" expression="^\*\*(\d+)$">
+ <action application="answer"/>
+ <action application="intercept" data="${db(select/${domain_name}-last_dial_ext/$1)}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="redial">
+ <condition field="destination_number" expression="^870$">
+ <action application="transfer" data="${db(select/${domain_name}-last_dial/${caller_id_number})}"/>
+ </condition>
+ </extension>
+
+ <extension name="global" continue="true">
+ <condition field="${network_addr}" expression="^$" break="never">
+ <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/>
+ <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/>
+ </condition>
+ <!-- This will setup some variables if the user isn't authenticated. -->
+ <condition field="${numbering_plan}" expression="^$" break="never">
+ <action application="set_user" data="default@${domain_name}"/>
+ </condition>
+ <condition field="$${call_debug}" expression="^true$" break="never">
+ <action application="info"/>
+ </condition>
+ <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
+ <action application="set" data="sip_secure_media=true"/>
+ <!-- Offer SRTP on outbound legs if we have it on inbound. -->
+ <!-- <action application="export" data="sip_secure_media=true"/> -->
+ </condition>
+ <condition>
+ <action application="db" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
+ <action application="db" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
+ <action application="db" data="insert/${domain_name}-last_dial/global/${uuid}"/>
+ </condition>
+ </extension>
+
+ <!-- If sip_req_host is not a local domain then this has to be an external sip uri -->
+ <extension name="external_sip_uri" continue="true">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="${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(.*)$|^\*0(.*)$">
+ <action application="answer"/>
+ <action application="eavesdrop" data="${db(select/${domain_name}-spymap/$1)}"/>
+ </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="${db(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="{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="set" data="call_timeout=10"/>
+ <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/>
+ </condition>
+ </extension>
+
+ <extension name="extension-intercom">
+ <!-- <condition field="${sip_to_params}" expression="intercom\=true"/> -->
+ <condition field="destination_number" expression="^8(10[01][0-9])$">
+ <action application="set" data="dialed_extension=$1"/>
+ <!-- This Alert-Info seems to be a case for Intercom for Polycom which sip_auto_answer=true covers already. -->
+ <!--<action application="export"><![CDATA[alert_info=<sip:${domain_name}>;Ring;Answer]]></action>-->
+ <action application="export"><![CDATA[sip_h_Call-Info=<sip:${domain_name}>;answer-after=0]]></action>
+ <action application="export" data="sip_invite_params=intercom=true"/>
+ <action application="export" data="sip_auto_answer=true"/>
+ <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
+ </condition>
+ </extension>
+
+ <!--
+ if the calling party is the called party, go to their VM
+ if the calling party is NOT the called party 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="set" data="dialed_extension=$1"/>
+ <action application="export" data="dialed_extension=$1"/>
+ </condition>
+ <condition field="destination_number" expression="^${caller_id_number}$">
+ <action application="set" data="voicemail_authorized=${sip_authorized}"/>
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="voicemail" data="check default ${domain_name} ${dialed_extension}"/>
+ <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
+ <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
+ <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+ <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
+ <anti-action application="set" data="ringback=${us-ring}"/>
+ <anti-action application="set" data="transfer_ringback=$${hold_music}"/>
+ <anti-action application="set" data="call_timeout=30"/>
+ <!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> -->
+ <anti-action application="set" data="hangup_after_bridge=true"/>
+ <!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
+ <anti-action application="set" data="continue_on_fail=true"/>
+ <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
+ <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
+ <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
+ <anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>
+ <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
+ <anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
+ <anti-action application="answer"/>
+ <anti-action application="sleep" data="1000"/>
+ <anti-action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
+ </condition>
+ </extension>
+
+ <!-- voicemail operator extension -->
+ <extension name="operator">
+ <condition field="destination_number" expression="^operator$|^0$">
+ <action application="set" data="transfer_ringback=$${hold_music}"/>
+ <action application="transfer" data="1000 XML features"/>
+ </condition>
+ </extension>
+
+ <!-- voicemail main extension -->
+ <extension name="vmain">
+ <condition field="destination_number" expression="^vmain|4000$">
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="voicemail" data="check default ${domain_name}"/>
+ </condition>
+ </extension>
+
+ <!-- dial via SIP uri -->
+ <extension name="sip_uri">
+ <condition field="destination_number" expression="^sip:(.*)$">
+ <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>
+
+ <!-- dial the freeswitch conference via SIP-->
+ <extension name="freeswitch_public_conf_via_sip">
+ <condition field="destination_number" expression="^9(888|1616)$">
+ <action application="bridge" data="sofia/${use_profile}/$1 at conference.freeswitch.org"/>
+ </condition>
+ </extension>
+
+ <!--This extension will start a conference and invite several people upon entering -->
+ <extension name="mad_boss">
+ <condition field="destination_number" expression="^0911$">
+
+ <!--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="sofia/gateway/$${default_provider}/19184238080"/>
+ <action application="conference_set_auto_outcall" data="sofia/default/888 at conference.freeswitch.org"/>
+
+ <action application="conference" data="cool at default"/>
+ </condition>
+ </extension>
+
+ <!-- a sample IVR -->
+ <extension name="ivr_demo">
+ <condition field="destination_number" expression="^5000$">
+ <action application="answer"/>
+ <action application="sleep" data="2000"/>
+ <action application="ivr" data="demo_ivr"/>
+ </condition>
+ </extension>
+
+ <!-- Create a conference on the fly and pull someone in at the same time. -->
+ <extension name="dyanmic conference">
+ <condition field="destination_number" expression="^5001$">
+ <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234 at conference.freeswitch.org"/>
+ </condition>
+ </extension>
+
+ <extension name="rtp_multicast_page">
+ <condition field="destination_number" expression="^pagegroup$|^7243">
+ <action application="answer"/>
+ <action application="esf_page_group"/>
+ </condition>
+ </extension>
+
+ <!--
+ Parking extensions... transferring calls to 5900 will park them in a queue.
+ -->
+ <extension name="park">
+ <condition field="destination_number" expression="^5900$">
+ <action application="set" data="fifo_music=$${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>
+
+ <!--
+ 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$">
+ <action application="pre_answer"/>
+ <action application="sleep" data="20000"/>
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="playback" data="voicemail/vm-goodbye.wav"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <!-- Send a 180 and let the far end generate ringback. -->
+ <extension name="ringback_180">
+ <condition field="destination_number" expression="^9980$">
+ <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="^9981$">
+ <action application="set" data="ringback=$${uk-ring}"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <!-- Send a 183 and use music as the ringtone. (early media) -->
+ <extension name="ringback_183_music_ring">
+ <condition field="destination_number" expression="^9982$">
+ <action application="set" data="ringback=$${hold_music}"/>
+ <action application="bridge" data="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="^9983$">
+ <action application="set" data="transfer_ringback=$${uk-ring}"/>
+ <action application="answer"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <!-- Answer the call and use music as the ringtone. (post answer) -->
+ <extension name="ringback_post_answer_music">
+ <condition field="destination_number" expression="^9984$">
+ <action application="set" data="transfer_ringback=$${hold_music}"/>
+ <action application="answer"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <extension name="show_info">
+ <condition field="destination_number" expression="^9992$">
+ <action application="answer"/>
+ <action application="info"/>
+ <action application="sleep" data="250"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <extension name="video_record">
+ <condition field="destination_number" expression="^9993$">
+ <action application="answer"/>
+ <action application="record_fsv" data="/tmp/testrecord.fsv"/>
+ </condition>
+ </extension>
+
+ <extension name="video_playback">
+ <condition field="destination_number" expression="^9994$">
+ <action application="answer"/>
+ <action application="play_fsv" data="/tmp/testrecord.fsv"/>
+ </condition>
+ </extension>
+
+ <extension name="delay_echo">
+ <condition field="destination_number" expression="^9995$">
+ <action application="answer"/>
+ <action application="delay_echo" data="5000"/>
+ </condition>
+ </extension>
+
+ <extension name="echo">
+ <condition field="destination_number" expression="^9996$">
+ <action application="answer"/>
+ <action application="echo"/>
+ </condition>
+ </extension>
+
+ <extension name="milliwatt">
+ <condition field="destination_number" expression="^9997$">
+ <action application="answer"/>
+ <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/>
+ </condition>
+ </extension>
+
+ <extension name="tone_stream">
+ <condition field="destination_number" expression="^9998$">
+ <action application="answer"/>
+ <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/>
+ </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="^9999$"/>
+ <condition field="${sip_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="$${hold_music}"/>
+ <anti-action application="answer"/>
+ <anti-action application="playback" data="$${hold_music}"/>
+ </condition>
+ </extension>
+
+<extension name="outgoing-fxo-channel-1">
+<!--
+ <condition field="destination_number" expression="^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">
+-->
+ <condition field="destination_number" expression="^([0-9][0-9][0-9][0-9][0-9][0-9])$">
+ <action application="set" data="dialed_ext=$1"/>
+ <action application="bridge" data="openzap/1/1/${dialed_ext}"/>
+ </condition>
+</extension>
+
+ <!--
+ You can place files in the default directory to get included.
+ -->
+ <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
+
+ <!--
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+
+ Anything you put below this line will usually get ignored due to the file in
+ default/99999_enum.xml as it will transfer the call to the enum dialplan.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+ -->
+
+ <!--
+ This is an example of how to override the RURI on an outgoing invite to a registered contact.
+ -->
+ <!--
+ <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>
+
+ <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>
+ -->
+
+ <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! -->
+
+ </context>
+</include>
Added: freeswitch/branches/gmaruzz/stuff/modules.conf.xml
==============================================================================
--- (empty file)
+++ freeswitch/branches/gmaruzz/stuff/modules.conf.xml Tue Nov 4 10:11:45 2008
@@ -0,0 +1,99 @@
+<configuration name="modules.conf" description="Modules">
+ <modules>
+
+ <!-- Loggers (I'd load these first) -->
+ <load module="mod_console"/>
+ <load module="mod_logfile"/>
+ <!-- <load module="mod_syslog"/> -->
+
+ <!--<load module="mod_yaml"/>-->
+
+ <!-- Multi-Faceted -->
+ <!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
+ <load module="mod_enum"/>
+
+ <!-- XML Interfaces -->
+ <!-- <load module="mod_xml_rpc"/> -->
+ <!-- <load module="mod_xml_curl"/> -->
+ <!-- <load module="mod_xml_cdr"/> -->
+
+ <!-- Event Handlers -->
+ <load module="mod_cdr_csv"/>
+ <!-- <load module="mod_event_multicast"/> -->
+ <load module="mod_event_socket"/>
+ <!-- <load module="mod_zeroconf"/> -->
+
+ <!-- Directory Interfaces -->
+ <!-- <load module="mod_ldap"/> -->
+
+ <!-- Endpoints -->
+ <!-- <load module="mod_dingaling"/> -->
+ <!-- <load module="mod_iax"/> -->
+ <load module="mod_portaudio"/>
+ <!--
+ <load module="mod_reference"/>
+ -->
+ <!-- <load module="mod_alsa"/> -->
+ <load module="mod_sofia"/>
+ <load module="mod_loopback"/>
+ <!-- <load module="mod_woomera"/> -->
+ <!-- <load module="mod_openzap"/> -->
+
+ <!-- Applications -->
+ <load module="mod_commands"/>
+ <load module="mod_conference"/>
+ <load module="mod_dptools"/>
+ <load module="mod_expr"/>
+ <load module="mod_fifo"/>
+ <load module="mod_voicemail"/>
+ <load module="mod_limit"/>
+ <load module="mod_esf"/>
+ <load module="mod_fsv"/>
+
+ <!-- SNOM Module -->
+ <!--<load module="mod_snom"/>-->
+
+ <!-- Dialplan Interfaces -->
+ <!-- <load module="mod_dialplan_directory"/> -->
+ <load module="mod_dialplan_xml"/>
+ <load module="mod_dialplan_asterisk"/>
+
+ <!-- Codec Interfaces -->
+ <load module="mod_voipcodecs"/>
+ <load module="mod_g723_1"/>
+ <load module="mod_g729"/>
+ <load module="mod_amr"/>
+ <load module="mod_ilbc"/>
+ <load module="mod_speex"/>
+ <load module="mod_h26x"/>
+
+ <!-- File Format Interfaces -->
+ <load module="mod_sndfile"/>
+ <load module="mod_native_file"/>
+ <!--For icecast/mp3 streams/files-->
+ <!--<load module="mod_shout"/>-->
+ <!--For local streams (play all the files in a directory)-->
+ <load module="mod_local_stream"/>
+ <load module="mod_tone_stream"/>
+
+ <!-- Timers -->
+
+ <!-- Languages -->
+ <load module="mod_spidermonkey"/>
+ <!-- <load module="mod_perl"/> -->
+ <!-- <load module="mod_python"/> -->
+ <!-- <load module="mod_java"/> -->
+ <load module="mod_lua"/>
+
+ <!-- ASR /TTS -->
+ <!-- <load module="mod_flite"/> -->
+ <!-- <load module="mod_pocketsphinx"/> -->
+ <!-- <load module="mod_cepstral"/> -->
+ <!-- <load module="mod_openmrcp"/> -->
+ <!-- <load module="mod_rss"/> -->
+
+ <!-- Say -->
+ <load module="mod_say_en"/>
+ <!-- <load module="mod_say_zh"/> -->
+ </modules>
+</configuration>
Added: freeswitch/branches/gmaruzz/stuff/openzap.conf.xml
==============================================================================
--- (empty file)
+++ freeswitch/branches/gmaruzz/stuff/openzap.conf.xml Tue Nov 4 10:11:45 2008
@@ -0,0 +1,50 @@
+<configuration name="openzap.conf" description="OpenZAP Configuration">
+ <settings>
+ <param name="debug" value="0"/>
+ <!--<param name="hold-music" value="$${moh_uri}"/>-->
+ <!--<param name="enable-analog-option" value="call-swap"/>-->
+ <!--<param name="enable-analog-option" value="3-way"/>-->
+ </settings>
+ <pri_spans>
+ <span name="PRI_1">
+ <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
+ <param name="q921loglevel" value="alert"/>
+ <param name="q931loglevel" value="alert"/>
+ <param name="mode" value="user"/>
+ <param name="dialect" value="5ess"/>
+ <param name="dialplan" value="XML"/>
+ <param name="context" value="default"/>
+ </span>
+ <span name="PRI_2">
+ <param name="q921loglevel" value="alert"/>
+ <param name="q931loglevel" value="alert"/>
+ <param name="mode" value="user"/>
+ <param name="dialect" value="5ess"/>
+ <param name="dialplan" value="XML"/>
+ <param name="context" value="default"/>
+ </span>
+ </pri_spans>
+ <!-- one entry here per openzap span -->
+ <analog_spans>
+ <span id="1">
+ <!--<param name="hold-music" value="$${moh_uri}"/>-->
+ <!--<param name="enable-analog-option" value="call-swap"/>-->
+ <!--<param name="enable-analog-option" value="3-way"/>-->
+ <param name="tonegroup" value="us"/>
+ <param name="digit-timeout" value="2000"/>
+ <param name="max-digits" value="11"/>
+ <param name="dialplan" value="XML"/>
+ <param name="context" value="default"/>
+ <param name="enable-callerid" value="true"/>
+ <!--
+ <param name="dial-regex" value="9996"/>
+ <param name="dial-regex-fail" value="9996"/>
+ -->
+
+ <!-- regex to stop dialing when it matches -->
+ <!--<param name="dial-regex" value="5555"/>-->
+ <!-- regex to stop dialing when it does not match -->
+ <!--<param name="fail-dial-regex" value="^5"/>-->
+ </span>
+ </analog_spans>
+</configuration>
Added: freeswitch/branches/gmaruzz/stuff/portaudio.conf.xml
==============================================================================
--- (empty file)
+++ freeswitch/branches/gmaruzz/stuff/portaudio.conf.xml Tue Nov 4 10:11:45 2008
@@ -0,0 +1,33 @@
+<configuration name="portaudio.conf" description="Soundcard Endpoint">
+ <settings>
+ <!-- indev, outdev, ringdev:
+ partial case sensitive string match on something in the name
+ or the device number prefixed with # eg "#1" (or blank for default) -->
+
+ <!-- device to use for input -->
+ <param name="indev" value="#7"/>
+ <!-- device to use for output -->
+ <param name="outdev" value="#7"/>
+
+ <!--device to use for inbound ring -->
+ <param name="ringdev" value="#7"/>
+ <!--File to play as the ring sound -->
+ <!--<param name="ring-file" value="/sounds/ring.wav"/>-->
+ <!--Number of seconds to pause between rings -->
+ <!--<param name="ring-interval" value="5"/>-->
+
+ <!--file to play when calls are on hold-->
+ <!--<param name="hold-file" value="/sounds/holdmusic.wav"/>-->
+ <!--Timer to use for hold music (i'd leave this one commented)-->
+ <!--<param name="timer-name" value="soft"/>-->
+
+ <!--Default dialplan and caller-id info -->
+ <param name="dialplan" value="XML"/>
+ <param name="cid-name" value="$${outbound_caller_name}"/>
+ <param name="cid-num" value="$${outbound_caller_id}"/>
+
+ <!--audio sample rate and interval -->
+ <param name="sample-rate" value="8000"/>
+ <param name="codec-ms" value="20"/>
+ </settings>
+</configuration>
Added: freeswitch/branches/gmaruzz/stuff/skypiax.conf.xml
==============================================================================
--- (empty file)
+++ freeswitch/branches/gmaruzz/stuff/skypiax.conf.xml Tue Nov 4 10:11:45 2008
@@ -0,0 +1,46 @@
+
+<configuration name="skypiax.conf" description="Skype Endpoint">
+ <settings>
+ <param name="debug" value="8"/>
+ <param name="port" value="4569"/>
+ <param name="ip" value="127.0.0.1"/>
+ <param name="codec-master" value="us"/>
+ <param name="dialplan" value="default"/>
+ <param name="codec-prefs" value="gsm,ulaw"/>
+ <param name="codec-rates" value="8000,16000"/>
+
+ <!-- PORTAUDIO BEGINS -->
+ <!-- indev, outdev, ringdev:
+ partial case sensitive string match on something in the name
+ or the device number prefixed with # eg "#1" (or blank for default) -->
+
+ <!-- device to use for input -->
+ <param name="indev" value="#7"/>
+ <!-- device to use for output -->
+ <param name="outdev" value="#7"/>
+
+ <!--device to use for inbound ring -->
+ <param name="ringdev" value="#7"/>
+ <!--File to play as the ring sound -->
+ <!--<param name="ring-file" value="/sounds/ring.wav"/>-->
+ <!--Number of seconds to pause between rings -->
+ <!--<param name="ring-interval" value="5"/>-->
+
+ <!--file to play when calls are on hold-->
+ <!--<param name="hold-file" value="/sounds/holdmusic.wav"/>-->
+ <!--Timer to use for hold music (i'd leave this one commented)-->
+ <!--<param name="timer-name" value="soft"/>-->
+
+ <!--Default dialplan and caller-id info -->
+ <!--
+ <param name="dialplan" value="XML"/>
+ -->
+ <param name="cid-name" value="$${outbound_caller_name}"/>
+ <param name="cid-num" value="$${outbound_caller_id}"/>
+
+ <!--audio sample rate and interval -->
+ <param name="sample-rate" value="8000"/>
+ <param name="codec-ms" value="20"/>
+ <!-- PORTAUDIO ENDS -->
+ </settings>
+</configuration>
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