429 lines
21 KiB
XML
429 lines
21 KiB
XML
<profile name="internal">
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<!--
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This is a sofia sip profile/user agent. This will service exactly one ip and port.
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In FreeSWITCH you can run multiple sip user agents on their own ip and port.
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When you hear someone say "sofia profile" this is what they are talking about.
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-->
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<!--aliases are other names that will work as a valid profile name for this profile-->
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<aliases>
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<!--
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<alias name="default"/>
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-->
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</aliases>
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<!-- Outbound Registrations -->
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<gateways>
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</gateways>
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<domains>
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<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
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<!--<domain name="$${domain}" parse="true"/>-->
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<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
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<!--<domain name="all" alias="true" parse="true"/>-->
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<domain name="all" alias="true" parse="false"/>
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</domains>
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<settings>
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<!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
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<!-- <param name="rtp-digit-delay" value="40"/>-->
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<!--
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When calls are in no media this will bring them back to media
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when you press the hold button.
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-->
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<!--<param name="media-option" value="resume-media-on-hold"/> -->
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<!--
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This will allow a call after an attended transfer go back to
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bypass media after an attended transfer.
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-->
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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 Rocks!"/> -->
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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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<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
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<!-- <param name="presence-proto-lookup" value="true"/> -->
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<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
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<!--<param name="liberal-dtmf" value="true"/>-->
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<!--
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Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
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responding. These options allow you to enable and control a watchdog
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on the Sofia SIP stack so that if it stops responding for the
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specified number of milliseconds, it will cause FreeSWITCH to crash
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immediately. This is useful if you run in an HA environment and
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need to ensure automated recovery from such a condition. Note that if
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your server is idle a lot, the watchdog may fire due to not receiving
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any SIP messages. Thus, if you expect your system to be idle, you
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should leave the watchdog disabled. It can be toggled on and off
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through the FreeSWITCH CLI either on an individual profile basis or
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globally for all profiles. So, if you run in an HA environment with a
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master and slave, you should use the CLI to make sure the watchdog is
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only enabled on the master.
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If such crash occurs, FreeSWITCH will dump core if allowed. The
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stacktrace will include function watchdog_triggered_abort().
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-->
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<param name="watchdog-enabled" value="no"/>
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<param name="watchdog-step-timeout" value="30000"/>
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<param name="watchdog-event-timeout" value="30000"/>
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<param name="log-auth-failures" value="false"/>
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<param name="forward-unsolicited-mwi-notify" value="false"/>
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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="$${internal_sip_port}"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
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<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
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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="nat.auto"/>
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<!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
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<!-- <param name="cid-in-1xx" value="false"/> -->
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<!-- extended info parsing -->
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<!-- <param name="extended-info-parsing" value="true"/> -->
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<!--<param name="aggressive-nat-detection" value="true"/>-->
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<!--
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There are known issues (asserts and segfaults) when 100rel is enabled.
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It is not recommended to enable 100rel at this time.
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-->
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<!--<param name="enable-100rel" value="true"/>-->
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<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
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<!-- RFC3263 Section 4.3 -->
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<!--<param name="disable-srv503" value="true"/>-->
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<!-- Enable Compact SIP headers. -->
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<!--<param name="enable-compact-headers" value="true"/>-->
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<!--
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enable/disable session timers
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-->
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<!--<param name="enable-timer" value="false"/>-->
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<!--<param name="minimum-session-expires" value="120"/>-->
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<param name="apply-inbound-acl" value="domains"/>
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<!--
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This defines your local network, by default we detect your local network
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and create this localnet.auto ACL for this.
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-->
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<param name="local-network-acl" value="localnet.auto"/>
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<!--<param name="apply-register-acl" value="domains"/>-->
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<!--<param name="dtmf-type" value="info"/>-->
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<!-- Added for Microsoft Edge support -->
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<param name="apply-candidate-acl" value="wan_v6.auto"/>
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<param name="apply-candidate-acl" value="rfc1918.auto"/>
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<param name="apply-candidate-acl" value="any_v6.auto"/>
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<param name="apply-candidate-acl" value="wan_v4.auto"/>
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<param name="apply-candidate-acl" value="any_v4.auto"/>
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<!-- 'true' means every time 'first-only' means on the first register -->
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<!--<param name="send-message-query-on-register" value="true"/>-->
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<!-- 'true' means every time 'first-only' means on the first register -->
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<!--<param name="send-presence-on-register" value="first-only"/> -->
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<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
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<!-- Remote-Party-ID header -->
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<!--<param name="caller-id-type" value="rpid"/>-->
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<!-- P-*-Identity family of headers -->
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<!--<param name="caller-id-type" value="pid"/>-->
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<!-- neither one -->
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<!--<param name="caller-id-type" value="none"/>-->
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<param name="record-path" value="$${recordings_dir}"/>
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<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
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<!--enable to use presence -->
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<param name="manage-presence" value="true"/>
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<!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
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<!--<param name="presence-probe-on-register" value="true"/>-->
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<!--<param name="manage-shared-appearance" value="true"/>-->
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<!-- used to share presence info across sofia profiles -->
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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="$${domain},$${local_ip_v4}"/>
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<param name="presence-privacy" value="$${presence_privacy}"/>
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<!-- ************************************************* -->
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<!-- This setting is for AAL2 bitpacking on G726 -->
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<!-- <param name="bitpacking" value="aal2"/> -->
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<!--max number of open dialogs in proceeding -->
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<!--<param name="max-proceeding" value="1000"/>-->
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<!--session timers for all call to expire after the specified seconds -->
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<!--<param name="session-timeout" value="1800"/>-->
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<!-- Can be 'true' or 'contact' -->
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<!--<param name="multiple-registrations" value="contact"/>-->
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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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<!--<param name="unregister-on-options-fail" value="true"/>-->
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<!-- Send an OPTIONS packet to all registered endpoints -->
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<!--<param name="all-reg-options-ping" value="true"/>-->
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<!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
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<!--<param name="nat-options-ping" value="true"/>-->
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<!--<param name="sip-options-respond-503-on-busy" value="true"/>-->
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<!--<param name="sip-messages-respond-200-ok" value="true"/>-->
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<!--<param name="sip-subscribe-respond-200-ok" value="true"/>-->
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${internal_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. (5061 will be used if unspecified) -->
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<param name="tls-sip-port" value="$${internal_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 'subjects_in', 'subjects_out' and 'subjects_all' 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 default: tlsv1,tlsv1.1,tlsv1.2 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH -->
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<param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
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<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
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(reduces delay on latent connections default true, must be disabled explicitly)-->
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<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
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<!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
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<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!-- Or, if you have PGSQL support, you can use that -->
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<!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!-- Let calls hit the dialplan before selecting codec for the a-leg -->
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<param name="inbound-late-negotiation" value="true"/>
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<!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
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<param name="inbound-zrtp-passthru" value="true"/>
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<!--TTL for nonce in sip auth-->
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<param name="nonce-ttl" value="60"/>
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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that the originator is using-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!-- Handle 302 Redirect in the dialplan -->
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<!--<param name="manual-redirect" value="true"/> -->
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<!-- Disable Transfer -->
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<!--<param name="disable-transfer" value="true"/> -->
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<!-- Disable Register -->
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<!--<param name="disable-register" value="true"/> -->
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<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
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<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
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<param name="auth-calls" value="$${internal_auth_calls}"/>
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<!-- Force the user and auth-user to match. -->
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<param name="inbound-reg-force-matching-username" value="true"/>
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<!-- on authed calls, authenticate *all* the packets not just invite -->
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<param name="auth-all-packets" value="false"/>
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<!-- external_sip_ip
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Used as the public IP address for SDP.
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Can be an one of:
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ip address - "12.34.56.78"
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a stun server lookup - "stun:stun.server.com"
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a DNS name - "host:host.server.com"
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auto - Use guessed ip.
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auto-nat - Use ip learned from NAT-PMP or UPNP
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-->
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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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<!-- 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="alias" value="sip:10.0.1.251:5555"/>-->
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<!--
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These are enabled to make the default config work better out of the box.
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If you need more than ONE domain you'll need to not use these options.
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-->
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<!--all inbound reg will look in this domain for the users -->
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<param name="force-register-domain" value="$${domain}"/>
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<!--force the domain in subscriptions to this value -->
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<param name="force-subscription-domain" value="$${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="$${domain}"/>
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<!-- for sip over websocket support -->
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<!-- param name="ws-binding" value=":5066"/ -->
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<!-- for sip over secure websocket support -->
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<!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
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<!-- param name="wss-binding" value=":7443"/ -->
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<!--<param name="delete-subs-on-register" value="false"/>-->
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<!-- launch a new thread to process each new inbound register when using heavier backends -->
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<!-- <param name="inbound-reg-in-new-thread" value="true"/> -->
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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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<!--force subscription expires to a lower value than requested-->
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<!--<param name="force-subscription-expires" value="60"/>-->
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<!-- add a random deviation to the expires value of the 202 Accepted -->
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<!--<param name="sip-subscription-max-deviation" value="120"/>-->
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<!-- disable register and transfer which may be undesirable in a public switch -->
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<!--<param name="disable-transfer" value="true"/>-->
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<!--<param name="disable-register" value="true"/>-->
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<!--
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enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
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right away, proxy waits until the call has been answered then sends accepts
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-->
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<param name="enable-3pcc" value="proxy"/>
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<!-- use at your own risk or if you know what this does.-->
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<!--<param name="NDLB-force-rport" value="true"/>-->
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<!--
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Choose the realm challenge key. Default is auto_to if not set.
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auto_from - uses the from field as the value for the sip realm.
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auto_to - uses the to field as the value for the sip realm.
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<anyvalue> - you can input any value to use for the sip realm.
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If you want URL dialing to work you'll want to set this to auto_from.
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If you use any other value besides auto_to or auto_from you'll
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loose the ability to do multiple domains.
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Note: comment out to restore the behavior before 2008-09-29
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-->
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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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<!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
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<!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
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<!-- on outbound calls set the callid to match the uuid of the session -->
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<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
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<!-- set to false disable this feature -->
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<!--<param name="rtp-autofix-timing" value="false"/>-->
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<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
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<!--<param name="pass-callee-id" value="false"/>-->
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<!-- clear clears them all or supply the name to add or the name
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prefixed with ~ to remove valid values:
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clear
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CISCO_SKIP_MARK_BIT_2833
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SONUS_SEND_INVALID_TIMESTAMP_2833
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-->
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<!--<param name="auto-rtp-bugs" data="clear"/>-->
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<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
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<!--<param name="disable-srv" value="false" />-->
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<!--<param name="disable-naptr" value="false" />-->
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<!-- The following can be used to fine-tune timers within sofia's transport layer
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Those settings are for advanced users and can safely be left as-is -->
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<!-- Initial retransmission interval (in milliseconds).
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Set the T1 retransmission interval used by the SIP transaction engine.
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The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
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<!-- <param name="timer-T1" value="500" /> -->
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<!-- Transaction timeout (defaults to T1 * 64).
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Set the T1x64 timeout value used by the SIP transaction engine.
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The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
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The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
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<!-- <param name="timer-T1X64" value="32000" /> -->
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<!-- Maximum retransmission interval (in milliseconds).
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Set the maximum retransmission interval used by the SIP transaction engine.
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The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
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Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
|
|
until the timer B fires. -->
|
|
<!-- <param name="timer-T2" value="4000" /> -->
|
|
|
|
<!--
|
|
Transaction lifetime (in milliseconds).
|
|
Set the lifetime for completed transactions used by the SIP transaction engine.
|
|
A completed transaction is kept around for the duration of T4 in order to catch late responses.
|
|
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
|
|
<!-- <param name="timer-T4" value="4000" /> -->
|
|
|
|
<!-- Turn on a jitterbuffer for every call -->
|
|
<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
|
|
|
|
|
|
<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
|
|
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
|
|
It's probably not what you want so stick with the default unless you really need to change this.
|
|
-->
|
|
<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
|
|
|
|
</settings>
|
|
</profile>
|