mirror of
https://github.com/signalwire/freeswitch.git
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Travis Cross
57abca11a8
This adds a parameter tls-ciphers in which we can specify the TLS cipher suite specification for OpenSSL.
498 lines
24 KiB
XML
498 lines
24 KiB
XML
<configuration name="sofia.conf" description="sofia endpoint">
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<global_settings>
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<param name="log-level" value="0"/>
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<param name="auto-restart" value="false"/>
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<param name="debug-presence" value="0"/>
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<!-- <param name="capture-server" value="udp:homer.example.com:5060"/> -->
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</global_settings>
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<profiles>
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<!--
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This is a sofia sip profile/user agent. This will service exactly one
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ip and port. In FreeSWITCH you can run multiple sip user agents on
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their own ip and port.
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-->
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<profile name="example">
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<gateways>
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<gateway name="example-gateway">
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<!-- account username (required) -->
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<param name="username" value="cluecon"/>
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<!-- auth realm (same as gateway name, if blank) -->
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<param name="realm" value="example.com"/>
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<!-- username to use in from (same as username, if blank) -->
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<param name="from-user" value="cluecon"/>
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<!-- domain to use in from (same as realm, if blank) /// -->
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<param name="from-domain" value="example.com"/>
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<!-- account password (required) -->
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<param name="password" value="xxxx"/>
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<!-- extension for inbound calls (same as username, if blank) -->
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<param name="extension" value="cluecon"/>
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<!-- proxy host (same as realm, if blank) -->
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<param name="proxy" value="example.com"/>
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<!-- send register to this proxy (same as proxy, if blank) -->
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<param name="register-proxy" value="example.com"/>
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<!-- expire in seconds (3600, if blank) -->
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<param name="expire-seconds" value="600"/>
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<!-- do not register -->
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<param name="register" value="false"/>
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<!-- which transport to use for register -->
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<param name="register-transport" value="tcp"/>
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<!-- how many seconds before a retry when a failure or timeout occurs
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-->
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<param name="retry-seconds" value="30"/>
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<!-- use the callerid of an inbound call in the from field on outbound
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calls via this gateway -->
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<param name="caller-id-in-from" value="false"/>
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<!-- extra sip params to send in the contact -->
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<param name="contact-params" value="tport=tcp"/>
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<!-- put the extension in the contact -->
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<param name="extension-in-contact" value="true"/>
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<!-- send an options ping every x seconds, failure will unregister
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and/or mark it down -->
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<param name="ping" value="25"/>
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<!-- callerid header mechanism -->
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<param name="cid-type" value="rpid"/>
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</gateway>
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</gateways>
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<aliases>
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<!-- aliases are other names that will work as a valid profile name for
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this profile -->
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<alias name="default"/>
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</aliases>
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<domains>
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<!-- indicator to parse the directory for domains with parse="true" to
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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
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get gateways and alias every domain to this profile -->
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<!-- This may seem like just a configuration convenience, but
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aliasing has real effects on the handling of packets. If
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you're sharing a profile between multiple realms, you
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want to alias all realms to the shared 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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<!-- When calls are in no media this will bring them back to media when
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you press the hold button. -->
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<!-- <param name="media-option" value="resume-media-on-hold"/> -->
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<!-- This will allow a call after an attended transfer go back to bypass
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media after an attended transfer. -->
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<!-- <param name="media-option" value="bypass-media-after-att-xfer"/> -->
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<!-- bypass again when hold is complete -->
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<!-- <param name="media-option" value="bypass-media-after-hold"/> -->
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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,
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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
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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
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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
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stop responding. These options allow you to enable and control a
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watchdog on the Sofia SIP stack so that if it stops responding for
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the specified number of milliseconds, it will cause FreeSWITCH to
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crash immediately. This is useful if you run in an HA environment
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and need to ensure automated recovery from such a condition. Note
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that if your server is idle a lot, the watchdog may fire due to not
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receiving any SIP messages. Thus, if you expect your system to be
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idle, you should leave the watchdog disabled. It can be toggled on
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and off through the FreeSWITCH CLI either on an individual profile
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basis or globally for all profiles. So, if you run in an HA
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environment with a master and slave, you should use the CLI to make
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sure the watchdog is 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
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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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<!-- There are known issues (asserts and segfaults) when 100rel is
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enabled. It is not recommended to enable 100rel at this time. -->
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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
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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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<!-- enable/disable session timers -->
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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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<!-- This defines your local network, by default we detect your local
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network and create this localnet.auto ACL for this. -->
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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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<!-- '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
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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
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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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<!-- 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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<!-- 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
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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
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unspecified) -->
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<param name="tls-sip-port" value="$${internal_tls_port}"/>
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<!-- Location of the agent.pem and cafile.pem ssl certificates (needed
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for TLS server) -->
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<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
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<!-- Optionally set the passphrase password used by openSSL to
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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
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servers (outbound) or handling inbound registration/invite requests
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how should we verify their certificate -->
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<!-- set to 'in' to only verify incoming connections, 'out' to only
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verify outgoing connections, 'all' to verify all connections, also
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'in_subjects', 'out_subjects' and 'all_subjects' for subject
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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
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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
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sets which subjects are allowed, multiple subjects can be split
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with a '|' pipe -->
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<param name="tls-verify-in-subjects" value=""/>
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<!-- Set the OpenSSL cipher suite list -->
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<!-- <param name="tls-ciphers" value="!aNULL:!LOW:!EXP:!kECDH:!ECDSA:!DSS:!PSK:!SRP:ALL"/> -->
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
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work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<!-- TLS maximum session lifetime -->
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<!-- <param name="tls-timeout" value="300"/> -->
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<!-- turn on auto-flush during bridge (skip timer sleep when the socket
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already has data) (reduces delay on latent connections default
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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
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another (on a per call basis with rtp_rewrite_timestamps chanvar)
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-->
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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
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of SQLite -->
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<!-- <param name="odbc-dsn" value="dsn:user:pass"/> -->
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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
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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
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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'
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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
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offer the codec 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
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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
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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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<!-- <param name="auth-messages" value="false"/> -->
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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
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box. If you need more than ONE domain you'll need to not use these
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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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<!-- <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
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rtcp_audio_interval_msec variable set to passthru to pass it across
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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 suscription 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
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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
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the call right away, proxy waits until the call has been answered
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then sends accepts
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-->
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<!-- <param name="enable-3pcc" value="true"/> -->
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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 loose
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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
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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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-->
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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-
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headers that it is supposed to ignore -->
|
|
<!-- <param name="pass-callee-id" value="false"/> -->
|
|
|
|
<!-- clear clears them all or supply the name to add or the name
|
|
prefixed with ~ to remove valid values:
|
|
|
|
clear
|
|
CISCO_SKIP_MARK_BIT_2833
|
|
SONUS_SEND_INVALID_TIMESTAMP_2833
|
|
|
|
-->
|
|
<!-- <param name="auto-rtp-bugs" data="clear"/> -->
|
|
|
|
<!-- the following can be used as workaround with bogus SRV/NAPTR
|
|
records -->
|
|
<!-- <param name="disable-srv" value="false" /> -->
|
|
<!-- <param name="disable-naptr" value="false" /> -->
|
|
|
|
<!-- The following can be used to fine-tune timers within sofia's
|
|
transport layer Those settings are for advanced users and can
|
|
safely be left as-is -->
|
|
|
|
<!-- Initial retransmission interval (in milliseconds).
|
|
|
|
Set the T1 retransmission interval used by the SIP transaction
|
|
engine.
|
|
|
|
The T1 is the initial duration used by request retransmission
|
|
timers A and E (UDP) as well as response retransmission timer G.
|
|
-->
|
|
<!-- <param name="timer-T1" value="500" /> -->
|
|
|
|
<!-- Transaction timeout (defaults to T1 * 64).
|
|
|
|
Set the T1x64 timeout value used by the SIP transaction engine.
|
|
|
|
The T1x64 is duration used for timers B, F, H, and J (UDP) by the
|
|
SIP transaction engine.
|
|
|
|
The timeout value T1x64 can be adjusted separately from the initial
|
|
retransmission interval T1. -->
|
|
<!-- <param name="timer-T1X64" value="32000" /> -->
|
|
|
|
|
|
<!-- Maximum retransmission interval (in milliseconds).
|
|
|
|
Set the maximum retransmission interval used by the SIP transaction
|
|
engine.
|
|
|
|
The T2 is the maximum duration used for the timers E (UDP) and G by
|
|
the SIP transaction engine.
|
|
|
|
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>
|
|
</profiles>
|
|
</configuration>
|