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freeswitch: SIP dial in support
This commit is contained in:
chandi
parent
4afa12bcf2
commit
e2ae542f29
@ -1,17 +0,0 @@
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<!--
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<extension name="from_my_provider">
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<condition field="destination_number" expression="^EXTERNALDID">
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<action application="answer"/>
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<action application="sleep" data="500"/>
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<action application="play_and_get_digits" data="5 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
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<action application="transfer" data="SEND_TO_CONFERENCE XML public"/>
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</condition>
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</extension>
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<extension name="check_if_conference_active">
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<condition field="${conference ${pin} list}" expression="/sofia/g" />
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<condition field="destination_number" expression="^SEND_TO_CONFERENCE$">
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<action application="set" data="bbb_authorized=true"/>
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<action application="transfer" data="${pin} XML default"/>
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</condition>
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</extension>
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-->
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@ -88,9 +88,12 @@ services:
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DISABLE_SOUND_ALONE: ${DISABLE_SOUND_ALONE:-false}
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SOUNDS_LANGUAGE: ${SOUNDS_LANGUAGE:-en-us-callie}
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ESL_PASSWORD: ${FSESL_PASSWORD:-ClueCon}
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{{ if .Env.SIP_IP_ALLOWLIST }}
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ports:
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- 5060:5060/udp
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{{ end }}
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volumes:
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- ./conf/sip_profiles:/etc/freeswitch/sip_profiles/external
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- ./conf/dialplan_public:/etc/freeswitch/dialplan/public_docker
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- ./conf/sip_profiles:/etc/freeswitch/sip_profiles/external-dialin
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- ./data/freeswitch-meetings:/var/freeswitch/meetings
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networks:
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bbb-net:
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29
mod/freeswitch/conf/dialplan/public/dialin.xml
Normal file
29
mod/freeswitch/conf/dialplan/public/dialin.xml
Normal file
@ -0,0 +1,29 @@
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<include>
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<extension name="from_my_provider">
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<condition field="destination_number" expression="^DIALIN">
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<action application="start_dtmf" />
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<action application="answer"/>
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<action application="sleep" data="1000"/>
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<action application="play_and_get_digits" data="9 9 3 30000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
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<action application="set_profile_var" data="caller_id_name=${regex(${caller_id_name}|^.*(.{4})$|xxx-xxx-%1)}"/>
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<action application="transfer" data="SEND_TO_CONFERENCE XML public"/>
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</condition>
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</extension>
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<extension name="check_if_conference_active">
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<condition field="${conference ${pin} list}" expression="/sofia/g" />
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<condition field="destination_number" expression="^SEND_TO_CONFERENCE$">
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<action application="set" data="bbb_authorized=true"/>
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<action application="transfer" data="${pin} XML default"/>
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</condition>
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</extension>
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<extension name="conf_bad_pin">
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<condition field="${pin}" expression="^\d{5}$">
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<action application="answer"/>
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<action application="sleep" data="1000"/>
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<action application="play_and_get_digits" data="9 9 3 30000 # conference/conf-bad-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
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<action application="transfer" data="SEND_TO_CONFERENCE XML public"/>
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</condition>
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</extension>
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</include>
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86
mod/freeswitch/conf/sip_profiles/external-dialin.xml
Normal file
86
mod/freeswitch/conf/sip_profiles/external-dialin.xml
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@ -0,0 +1,86 @@
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<profile name="external-dialin">
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<!-- This profile is only for outbound registrations to providers -->
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<gateways>
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<X-PRE-PROCESS cmd="include" data="external-dialin/*.xml"/>
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</gateways>
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<aliases>
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<!--
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<alias name="outbound"/>
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<alias name="nat"/>
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-->
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</aliases>
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<domains>
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<domain name="all" alias="false" parse="true"/>
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</domains>
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<settings>
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<param name="debug" value="1"/>
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<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
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<!-- <param name="shutdown-on-fail" value="true"/> -->
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<param name="sip-trace" value="no"/>
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<param name="sip-capture" value="no"/>
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<param name="rfc2833-pt" value="101"/>
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<!-- RFC 5626 : Send reg-id and sip.instance -->
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<!--<param name="enable-rfc-5626" value="true"/> -->
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
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<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
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<param name="hold-music" value="$${hold_music}"/>
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<param name="rtp-timer-name" value="soft"/>
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<!--<param name="enable-100rel" value="true"/>-->
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<!--<param name="disable-srv503" value="true"/>-->
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<!-- This could be set to "passive" -->
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="manage-presence" value="false"/>
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<!-- Added for Microsoft Edge browser -->
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<param name="apply-candidate-acl" value="localnet.auto"/>
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<param name="apply-candidate-acl" value="wan_v4.auto"/>
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<param name="apply-candidate-acl" value="rfc1918.auto"/>
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<param name="apply-candidate-acl" value="any_v4.auto"/>
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<!-- used to share presence info across sofia profiles
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manage-presence needs to be set to passive on this profile
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if you want it to behave as if it were the internal profile
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for presence.
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-->
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<!-- Name of the db to use for this profile -->
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<param name="dbname" value="sqlite://memory://file:external_dialin?mode=memory&cache=shared"/>
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<!--<param name="presence-hosts" value="$${domain}"/>-->
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<!--<param name="force-register-domain" value="$${domain}"/>-->
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<!--all inbound reg will stored in the db using this domain -->
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<!--<param name="force-register-db-domain" value="$${domain}"/>-->
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<!-- ************************************************* -->
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<!--<param name="aggressive-nat-detection" value="true"/>-->
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<param name="inbound-codec-negotiation" value="generous"/>
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<param name="nonce-ttl" value="60"/>
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<param name="auth-calls" value="false"/>
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<param name="inbound-late-negotiation" value="true"/>
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<param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="ext-rtp-ip" value="$${external_ip_v4}"/>
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<param name="ext-sip-ip" value="$${external_ip_v4}"/>
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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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<param name="enable-3pcc" value="proxy"/>
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<!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
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<param name="rtcp-audio-interval-msec" value="5000"/>
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<param name="rtcp-video-interval-msec" value="5000"/>
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<!-- Cut down in the join time -->
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<param name="dtmf-type" value="info"/>
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<param name="liberal-dtmf" value="true"/>
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</settings>
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</profile>
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115
mod/freeswitch/conf/sip_profiles/external.xml
Normal file
115
mod/freeswitch/conf/sip_profiles/external.xml
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@ -0,0 +1,115 @@
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<profile name="external">
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<!-- This profile is only for outbound registrations to providers -->
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<domains>
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<domain name="all" alias="false" parse="true"/>
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</domains>
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<settings>
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<param name="debug" value="0"/>
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<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
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<!-- <param name="shutdown-on-fail" value="true"/> -->
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<param name="sip-trace" value="no"/>
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<param name="sip-capture" value="no"/>
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<param name="rfc2833-pt" value="101"/>
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<!-- RFC 5626 : Send reg-id and sip.instance -->
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<!--<param name="enable-rfc-5626" value="true"/> -->
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<!--
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SIP port is not rquired, since we are using WS for the
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internal connection and a seperate profile (external-dialin-xml)
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for SIP dial in
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-->
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<param name="sip-port" value="15060"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
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<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
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<param name="hold-music" value="$${hold_music}"/>
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<param name="rtp-timer-name" value="soft"/>
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<!--<param name="enable-100rel" value="true"/>-->
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<!--<param name="disable-srv503" value="true"/>-->
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<!-- This could be set to "passive" -->
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="manage-presence" value="false"/>
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<!-- Added for Microsoft Edge browser -->
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<param name="apply-candidate-acl" value="localnet.auto"/>
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<param name="apply-candidate-acl" value="wan_v4.auto"/>
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<param name="apply-candidate-acl" value="rfc1918.auto"/>
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<param name="apply-candidate-acl" value="any_v4.auto"/>
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<!-- used to share presence info across sofia profiles
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manage-presence needs to be set to passive on this profile
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if you want it to behave as if it were the internal profile
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for presence.
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-->
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<!-- Name of the db to use for this profile -->
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<param name="dbname" value="sqlite://memory://file:external?mode=memory&cache=shared"/>
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<!--<param name="presence-hosts" value="$${domain}"/>-->
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<!--<param name="force-register-domain" value="$${domain}"/>-->
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<!--all inbound reg will stored in the db using this domain -->
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<!--<param name="force-register-db-domain" value="$${domain}"/>-->
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<!-- ************************************************* -->
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<!--<param name="aggressive-nat-detection" value="true"/>-->
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<param name="inbound-codec-negotiation" value="generous"/>
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<param name="nonce-ttl" value="60"/>
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<param name="auth-calls" value="false"/>
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<param name="inbound-late-negotiation" value="true"/>
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<param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
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<!--
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DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="ext-rtp-ip" value="auto-nat"/>
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<param name="ext-sip-ip" value="auto-nat"/>
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-->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
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<param name="ext-sip-ip" value="$${local_ip_v4}"/>
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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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<param name="enable-3pcc" value="proxy"/>
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${external_ssl_enable}"/>
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<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
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<param name="tls-only" value="false"/>
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<!-- additional bind parameters for TLS -->
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<param name="tls-bind-params" value="transport=tls"/>
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<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
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<param name="tls-sip-port" value="$${external_tls_port}"/>
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<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
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<!--<param name="tls-cert-dir" value=""/>-->
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<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
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<param name="tls-passphrase" value=""/>
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<!-- Verify the date on TLS certificates -->
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<param name="tls-verify-date" value="true"/>
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<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
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<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
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<param name="tls-verify-policy" value="none"/>
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<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
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<param name="tls-verify-depth" value="2"/>
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<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
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<param name="tls-verify-in-subjects" value=""/>
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<param name="ws-binding" value=":5066"/>
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<param name="wss-binding" value=":7443"/>
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<!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
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<param name="rtcp-audio-interval-msec" value="5000"/>
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<param name="rtcp-video-interval-msec" value="5000"/>
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<!-- Cut down in the join time -->
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<param name="dtmf-type" value="info"/>
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<param name="liberal-dtmf" value="true"/>
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</settings>
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</profile>
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@ -61,6 +61,7 @@
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information on this topic.
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-->
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<X-PRE-PROCESS cmd="set" data="local_ip_v4=10.7.7.10"/>
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<X-PRE-PROCESS cmd="set" data="external_ip_v4={{ .Env.EXTERNAL_IPv4 }}"/>
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<X-PRE-PROCESS cmd="set" data="domain={{ .Env.DOMAIN }}"/>
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<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
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@ -54,6 +54,7 @@ docker run \
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-e DEV_MODE=${DEV_MODE:-false} \
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-e IGNORE_TLS_CERT_ERRORS=${IGNORE_TLS_CERT_ERRORS:-} \
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-e EXTERNAL_IPv6=${EXTERNAL_IPv6:-} \
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-e SIP_IP_ALLOWLIST=${SIP_IP_ALLOWLIST:-} \
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-e ENABLE_RECORDING=${ENABLE_RECORDING:-false} \
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-e ENABLE_HTTPS_PROXY=${ENABLE_HTTPS_PROXY:-false} \
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-e ENABLE_WEBHOOKS=${ENABLE_WEBHOOKS:-false} \
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