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<profile name= "external" >
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!-- This profile is only for outbound registrations to providers -->
<gateways >
<X-PRE-PROCESS cmd= "include" data= "external/*.xml" />
</gateways>
<aliases >
<!--
<alias name= "outbound" />
<alias name= "nat" />
-->
</aliases>
<domains >
<domain name= "all" alias= "false" parse= "true" />
</domains>
<settings >
<param name= "debug" value= "0" />
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown - on - fail" value="true"/> -->
<param name= "sip-trace" value= "no" />
<param name= "sip-capture" value= "no" />
<param name= "rfc2833-pt" value= "101" />
<!-- RFC 5626 : Send reg - id and sip.instance -->
<!-- <param name="enable - rfc - 5626" value="true"/> -->
<param name= "sip-port" value= "$${external_sip_port}" />
<param name= "dialplan" value= "XML" />
<param name= "context" value= "public" />
<param name= "dtmf-duration" value= "2000" />
<param name= "inbound-codec-prefs" value= "$${global_codec_prefs}" />
<param name= "outbound-codec-prefs" value= "$${outbound_codec_prefs}" />
<param name= "hold-music" value= "$${hold_music}" />
<param name= "rtp-timer-name" value= "soft" />
<!-- <param name="enable - 100rel" value="true"/> -->
<!-- <param name="disable - srv503" value="true"/> -->
<!-- This could be set to "passive" -->
<param name= "local-network-acl" value= "localnet.auto" />
<param name= "manage-presence" value= "false" />
<!-- Added for Microsoft Edge browser -->
<param name= "apply-candidate-acl" value= "localnet.auto" />
<param name= "apply-candidate-acl" value= "wan_v4.auto" />
<param name= "apply-candidate-acl" value= "rfc1918.auto" />
<param name= "apply-candidate-acl" value= "any_v4.auto" />
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
-->
<!-- Name of the db to use for this profile -->
<!-- <param name="dbname" value="share_presence"/> -->
<!-- <param name="presence - hosts" value="$${domain}"/> -->
<!-- <param name="force - register - domain" value="$${domain}"/> -->
<!-- all inbound reg will stored in the db using this domain -->
<!-- <param name="force - register - db - domain" value="$${domain}"/> -->
<!-- ************************************************* -->
<!-- <param name="aggressive - nat - detection" value="true"/> -->
<param name= "inbound-codec-negotiation" value= "generous" />
<param name= "nonce-ttl" value= "60" />
<param name= "auth-calls" value= "false" />
<param name= "inbound-late-negotiation" value= "true" />
<param name= "inbound-zrtp-passthru" value= "true" /> <!-- (also enables late negotiation) -->
<!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
<param name= "rtp-ip" value= "$${local_ip_v4}" />
<param name= "sip-ip" value= "$${local_ip_v4}" />
<param name= "ext-rtp-ip" value= "auto-nat" />
<param name= "ext-sip-ip" value= "auto-nat" />
-->
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<param name= "rtp-ip" value= "0.0.0.0" />
<param name= "sip-ip" value= "0.0.0.0" />
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<param name= "ext-rtp-ip" value= "$${external_rtp_ip}" />
<param name= "ext-sip-ip" value= "$${external_sip_ip}" />
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<!--
Listen only clients somehow run into this timeout
causing
Hangup sofia/external/GLOBAL_AUDIO_76116@10.7.7.1 [CS_EXECUTE] [MEDIA_TIMEOUT]
[mcs-freeswitch] Dispatching conference new video floor event released
[mcs-freeswitch] Received CHANNEL_HANGUP for
-->
<param name= "rtp-timeout-sec" value= "86400" />
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<param name= "rtp-hold-timeout-sec" value= "1800" />
<param name= "enable-3pcc" value= "proxy" />
<!-- TLS: disabled by default, set to "true" to enable -->
<param name= "tls" value= "$${external_ssl_enable}" />
<!-- Set to true to not bind on the normal sip - port but only on the TLS port -->
<param name= "tls-only" value= "false" />
<!-- additional bind parameters for TLS -->
<param name= "tls-bind-params" value= "transport=tls" />
<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
<param name= "tls-sip-port" value= "$${external_tls_port}" />
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<!-- <param name="tls - cert - dir" value=""/> -->
<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
<param name= "tls-passphrase" value= "" />
<!-- Verify the date on TLS certificates -->
<param name= "tls-verify-date" value= "true" />
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name= "tls-verify-policy" value= "none" />
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name= "tls-verify-depth" value= "2" />
<!-- If the tls - verify - policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
<param name= "tls-verify-in-subjects" value= "" />
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name= "tls-version" value= "$${sip_tls_version}" />
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<param name= "ws-binding" value= "0.0.0.0:5066" />
<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 -->
<param name= "rtcp-audio-interval-msec" value= "5000" />
<param name= "rtcp-video-interval-msec" value= "5000" />
<!-- Cut down in the join time -->
<param name= "dtmf-type" value= "info" />
<param name= "liberal-dtmf" value= "true" />
</settings>
</profile>