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Sync up the external-ipv6 SIP Profile

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FusionPBX 2022-08-06 17:01:16 -06:00 committed by GitHub
parent eea25a336d
commit 6299e91d76
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1 changed files with 26 additions and 19 deletions
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45
resources/templates/conf/sip_profiles/external-ipv6.xml.noload
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@ -15,8 +15,8 @@
<settings>
<param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<param name="user-agent-string" value="FreeSWITCH" enabled="true"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<param name="shutdown-on-fail" value="true" enabled="false"/>
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
@ -25,14 +25,11 @@
<param name="enable-rfc-5626" value="true" enabled="false"/>
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="track-calls" value="false"/>
<param name="context" value="public"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="zrtp-passthru" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="enable-100rel" value="true" enabled="false"/>
<param name="disable-srv503" value="true" enabled="false"/>
@ -41,9 +38,9 @@
<param name="manage-presence" value="false"/>
<!-- 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.
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" enabled="false"/>
@ -58,13 +55,14 @@
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="auth-subscriptions" value="true"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<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="$${local_ip_v6}"/>
<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="sip-ip" value="$${local_ip_v6}"/>
<!-- Shouldn't set these on IPv6 -->
<param name="ext-rtp-ip" value="$${external_rtp_ip}" enabled="false"/>
<param name="ext-sip-ip" value="$${external_sip_ip}" enabled="false"/>
<param name="rtp-timeout-sec" value="300" description="deprecated" enabled="false"/>
@ -73,11 +71,6 @@
<param name="media_hold_timeout" value="1800" enabled="true"/>
<param name="enable-3pcc" value="true" enabled="false"/>
<!--session timers -->
<param name="session-timeout" value="0" enabled="true"/>
<param name="enable-timer" value="false" enabled="true"/>
<param name="minimum-session-expires" value="0" enabled="false"/>
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${external_ssl_enable}"/>
<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
@ -93,8 +86,8 @@
<!-- Verify the date on TLS certificates -->
<param name="tls-verify-date" value="false"/>
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="all|subjects_all" enabled="false"/>
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'subjects_in', 'subjects_out' and 'subjects_all' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="none"/>
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name="tls-verify-depth" value="2"/>
<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
@ -102,7 +95,21 @@
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<param name="zrtp-passthru" value="true"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!--session timers -->
<param name="session-timeout" value="0" enabled="true"/>
<param name="enable-timer" value="false" enabled="true"/>
<param name="minimum-session-expires" value="0" enabled="false"/>
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<param name="odbc-dsn" value="$${dsn}" enabled="false"/>
<param name="dtmf-type" value="rfc2833"/>
<!-- save session in the database for option to restore SIP UDP calls -->
<param name="track-calls" value="false"/>
</settings>
</profile>