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129 lines
6.2 KiB
XML
129 lines
6.2 KiB
XML
<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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<gateways>
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<X-PRE-PROCESS cmd="include" data="external/*.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="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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<param name="sip-port" value="$${external_sip_port}"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="$${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="none"/>
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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="share_presence"/>-->
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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="$${local_ip_v4}"/>
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<param name="ext-rtp-ip" value="auto-nat"/>
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<param name="ext-sip-ip" value="auto-nat"/>
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-->
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<param name="rtp-ip" value="$${external_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_rtp_ip}"/>
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<param name="ext-sip-ip" value="$${external_sip_ip}"/>
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<!--
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Listen only clients somehow run into this timeout
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causing
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Hangup sofia/external/GLOBAL_AUDIO_76116@10.7.7.1 [CS_EXECUTE] [MEDIA_TIMEOUT]
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[mcs-freeswitch] Dispatching conference new video floor event released
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[mcs-freeswitch] Received CHANNEL_HANGUP for
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-->
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<param name="rtp-timeout-sec" value="86400"/>
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<param name="rtp-hold-timeout-sec" value="1800"/>
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<param name="enable-3pcc" value="proxy"/>
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${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="0.0.0.0:5066"/>
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<param name="wss-binding" value="$${local_ip_v4}: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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