forked from Mirrors/freeswitch
91a6fc82c0
FS-7338: remove libsilk from tree, use system lib instead FS-7338: change to always use system liblua FS-7338: remove libbroadvoice from tree, use system lib instead FS-7338: remove libilbc from tree, use system lib instead FS-7338: remove libs using system libs from bootstrap FS-7338: remove libg722_1 from tree, use system lib instead FS-7338: remove mod_celt, it has be superseded by mod_opus FS-7338: remove libcodec2 from tree, use system lib instead FS-7338: remove libopus from tree, use system lib instead FS-7338: remove libsoundtouch build from tree, use system lib instead FS-7338: remove flite build from tree, use system lib instead FS-7338: remove openldap build from tree, use system lib instead FS-7338: remove libmongoc build from tree, use system lib instead FS-7338: remove mod_mongo deps that are no longer actually required FS-7338: remove some dup demo modules and don't include demo code in packages
280 lines
11 KiB
XML
280 lines
11 KiB
XML
<?xml version="1.0"?>
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<document type="freeswitch/xml">
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<X-PRE-PROCESS cmd="set" data="auto_answer=false"/>
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<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
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<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
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<X-PRE-PROCESS cmd="set" data="codec_prefs=G722,PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
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<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
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<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
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<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
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<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
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<X-PRE-PROCESS cmd="set" data="default_gateway=default"/>
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<X-PRE-PROCESS cmd="set" data="us-ring=%(2000, 4000, 440.0, 480.0)"/>
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<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
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<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
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<section name="configuration" description="Various Configuration">
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<configuration name="cdr_csv.conf" description="CDR CSV Format">
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<settings>
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<param name="default-template" value="example"/>
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<param name="rotate-on-hup" value="true"/>
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<param name="legs" value="a"/>
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</settings>
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<templates>
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<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
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</templates>
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</configuration>
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<configuration name="console.conf" description="Console Logger">
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<mappings>
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<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
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</mappings>
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<settings>
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<param name="colorize" value="true"/>
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<param name="loglevel" value="$${console_loglevel}"/>
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</settings>
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</configuration>
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<configuration name="enum.conf" description="ENUM Module">
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<settings>
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<param name="default-root" value="e164.org"/>
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<param name="default-isn-root" value="freenum.org"/>
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<param name="query-timeout" value="10"/>
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<param name="auto-reload" value="true"/>
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</settings>
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<routes>
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<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/softphone/$1"/>
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</routes>
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</configuration>
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<configuration name="local_stream.conf" description="stream files from local dir">
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<directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
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<param name="rate" value="48000"/>
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<param name="shuffle" value="true"/>
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<param name="channels" value="1"/>
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<param name="interval" value="10"/>
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<param name="timer-name" value="soft"/>
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</directory>
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</configuration>
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<configuration name="logfile.conf" description="File Logging">
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<settings>
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<param name="rotate-on-hup" value="true"/>
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</settings>
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<profiles>
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<profile name="default">
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<settings>
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</settings>
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<mappings>
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<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
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</mappings>
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</profile>
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</profiles>
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</configuration>
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<configuration name="modules.conf" description="Modules">
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<modules>
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<load module="mod_console"/>
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<load module="mod_logfile"/>
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<load module="mod_enum"/>
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<load module="mod_cdr_csv"/>
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<load module="mod_portaudio"/>
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<load module="mod_sofia"/>
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<load module="mod_loopback"/>
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<load module="mod_commands"/>
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<load module="mod_dptools"/>
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<load module="mod_dialplan_xml"/>
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<load module="mod_voipcodecs"/>
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<load module="mod_sndfile"/>
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<load module="mod_tone_stream"/>
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<load module="mod_local_stream"/>
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</modules>
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</configuration>
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<configuration name="portaudio.conf" description="Soundcard Endpoint">
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<settings>
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<param name="indev" value=""/>
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<!-- device to use for output -->
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<param name="outdev" value=""/>
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<!--<param name="ringdev" value=""/>-->
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<param name="ring-file" value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
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<param name="ring-interval" value="5"/>
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<param name="hold-file" value="$${hold_music}"/>
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<!--<param name="timer-name" value="soft"/>-->
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<param name="dialplan" value="XML"/>
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<param name="cid-name" value="$${outbound_caller_name}"/>
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<param name="cid-num" value="$${outbound_caller_number}"/>
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<param name="sample-rate" value="48000"/>
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<param name="codec-ms" value="10"/>
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</settings>
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</configuration>
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<configuration name="post_load_modules.conf" description="Modules">
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<modules>
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</modules>
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</configuration>
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<configuration name="sofia.conf" description="sofia Endpoint">
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<global_settings>
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<param name="log-level" value="0"/>
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<param name="auto-restart" value="true"/>
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<param name="debug-presence" value="0"/>
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</global_settings>
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<profiles>
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<profile name="softphone">
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<gateways>
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<X-PRE-PROCESS cmd="include" data="accounts/*.xml"/>
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</gateways>
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<settings>
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<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
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<!-- Can be set to "_undef_" to remove the User-Agent header -->
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<param name="user-agent-string" value="FreeSWITCH/SoftPhone"/>
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<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
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<param name="debug" value="0"/>
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<param name="sip-trace" value="no"/>
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<param name="context" value="public"/>
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<param name="rfc2833-pt" value="101"/>
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<!-- port to bind to for sip traffic -->
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<param name="sip-port" value="12345"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="$${codec_prefs}"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="hold-music" value="$${hold_music}"/>
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<param name="apply-nat-acl" value="rfc1918"/>
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<!--<param name="enable-timer" value="false"/>-->
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<!--<param name="enable-100rel" value="true"/>-->
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<!--<param name="minimum-session-expires" value="120"/>-->
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<!--<param name="dtmf-type" value="info"/>-->
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<param name="manage-presence" value="false"/>
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<!--<param name="bitpacking" value="aal2"/> -->
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<param name="max-proceeding" value="3"/>
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<!--<param name="session-timeout" value="120"/>-->
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<!--set to 'greedy' if you want your codec list to take precedence -->
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<param name="inbound-codec-negotiation" value="generous"/>
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<!-- if you want to send any special bind params of your own -->
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<!--<param name="bind-params" value="transport=udp"/>-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!--<param name="accept-blind-reg" value="true"/> -->
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<!--<param name="accept-blind-auth" value="true"/> -->
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<!--<param name="suppress-cng" value="true"/> -->
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<param name="nonce-ttl" value="60"/>
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
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<param name="auth-calls" value="false"/>
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<param name="auth-all-packets" value="false"/>
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<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
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<param name="ext-sip-ip" value="$${external_sip_ip}"/>
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<!-- rtp inactivity timeout -->
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<param name="rtp-timeout-sec" value="300"/>
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<param name="rtp-hold-timeout-sec" value="1800"/>
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<param name="disable-register" value="true"/>
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<!--<param name="NDLB-force-rport" value="true"/>-->
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<param name="challenge-realm" value="auto_from"/>
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<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
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<!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
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<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
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<!--<param name="auto-rtp-bugs" data="clear"/>-->
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</settings>
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</profile>
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</profiles>
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</configuration>
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<configuration name="switch.conf" description="Core Configuration">
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<cli-keybindings>
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<key name="1" value="help"/>
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<key name="2" value="status"/>
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<key name="3" value="pa answer"/>
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<key name="4" value="pa hangup"/>
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<key name="5" value="sofia status"/>
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<key name="6" value="reloadxml"/>
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<key name="7" value="console loglevel 0"/>
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<key name="8" value="console loglevel 7"/>
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<key name="9" value="sofia status profile softphone"/>
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<key name="10" value="fsctl pause"/>
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<key name="11" value="fsctl resume"/>
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<key name="12" value="version"/>
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</cli-keybindings>
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<settings>
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<param name="colorize-console" value="true"/>
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<param name="max-sessions" value="20"/>
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<param name="sessions-per-second" value="5"/>
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<param name="loglevel" value="debug"/>
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<param name="crash-protection" value="false"/>
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<param name="dump-cores" value="yes"/>
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<param name="rtp-start-port" value="16384"/>
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<param name="rtp-end-port" value="16484"/>
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</settings>
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</configuration>
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</section>
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<section name="dialplan" description="Regex/XML Dialplan">
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<context name="default">
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<extension name="codec_and_sip_uri">
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<condition field="destination_number" expression="^sip:(.*):(.*)$">
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<action application="bridge" data="{absolute_codec_string=$1}sofia/softphone/$2"/>
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</condition>
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</extension>
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<extension name="sip_uri">
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<condition field="destination_number" expression="^sip:(.*)$">
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<action application="bridge" data="sofia/softphone/$1"/>
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</condition>
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</extension>
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<extension name="codec_and_number">
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<condition field="destination_number" expression="^(.*):(.*)@(.*)$">
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<action application="bridge" data="{absolute_codec_string=$1}sofia/gateway/$3/$2"/>
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</condition>
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</extension>
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<extension name="number">
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<condition field="destination_number" expression="^(.*)@(.*)$">
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<action application="bridge" data="sofia/gateway/$2/$1"/>
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</condition>
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</extension>
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<extension name="number">
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<condition field="destination_number" expression="^(.*)$">
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<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
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</condition>
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</extension>
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</context>
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<context name="public">
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<extension name="public_extensions">
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<condition field="$${auto_answer}" expression="^true$"/>
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<condition field="destination_number" expression="^(.*)$">
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<action application="info"/>
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<action application="bridge" data="portaudio/auto_answer"/>
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</condition>
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</extension>
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<extension name="public_extensions">
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<condition field="${sip_to_params}" expression="intercom=true"/>
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<condition field="${alert_info}" expression="Ring;Answer"/>
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<condition field="destination_number" expression="^(.*)$">
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<action application="info"/>
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<action application="bridge" data="portaudio/auto_answer"/>
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</condition>
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</extension>
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<extension name="public_extensions">
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<condition field="destination_number" expression="^(.*)$">
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<action application="info"/>
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<action application="set" data="ringback=${us-ring}"/>
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<action application="pre_answer"/>
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<action application="bridge" data="portaudio"/>
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</condition>
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</extension>
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</context>
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</section>
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</document>
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