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}/$1@conference.freeswitch.org"/> + </condition> + </extension> + + <!--This extension will start a conference and invite several people upon entering --> + <extension name="mad_boss"> + <condition field="destination_number" expression="^0911$"> + + <!--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/$${default_provider}/19184238080"/> + <action application="conference_set_auto_outcall" data="sofia/default/888@conference.freeswitch.org"/> + + <action application="conference" data="cool@default"/> + </condition> + </extension> + + <!-- a sample IVR --> + <extension name="ivr_demo"> + <condition field="destination_number" expression="^5000$"> + <action application="answer"/> + <action application="sleep" data="2000"/> + <action application="ivr" data="demo_ivr"/> + </condition> + </extension> + + <!-- Create a conference on the fly and pull someone in at the same time. --> + <extension name="dyanmic conference"> + <condition field="destination_number" expression="^5001$"> + <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> + </condition> + </extension> + + <extension name="rtp_multicast_page"> + <condition field="destination_number" expression="^pagegroup$|^7243"> + <action application="answer"/> + <action application="esf_page_group"/> + </condition> + </extension> + + <!-- + Parking extensions... transferring calls to 5900 will park them in a queue. + --> + <extension name="park"> + <condition field="destination_number" expression="^5900$"> + <action application="set" data="fifo_music=$${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>