forked from Mirrors/freeswitch
322 lines
15 KiB
XML
322 lines
15 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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<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
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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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<!--
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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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<!-- <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="log-auth-failures" value="true"/>
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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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<!-- 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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<!-- '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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<!-- 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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<!--<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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<!-- ************************************************* -->
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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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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${internal_ssl_enable}"/>
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<!-- additional bind parameters for TLS -->
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<param name="tls-bind-params" value="transport=tls"/>
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<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
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<param name="tls-sip-port" value="$${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="$${internal_ssl_dir}"/>
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<!-- 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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<!--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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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" 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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<!--<param name="delete-subs-on-register" value="false"/>-->
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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 suscription expires to a lower value than requested-->
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<!--<param name="force-subscription-expires" value="60"/>-->
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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="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 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 prefixed with ~ to remove
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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
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until the timer B fires. -->
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<!-- <param name="timer-T2" value="4000" /> -->
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<!--
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Transaction lifetime (in milliseconds).
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Set the lifetime for completed transactions used by the SIP transaction engine.
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A completed transaction is kept around for the duration of T4 in order to catch late responses.
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The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
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<!-- <param name="timer-T4" value="4000" /> -->
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</settings>
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</profile>
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