forked from Mirrors/freeswitch
205cf0534f
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9934 d0543943-73ff-0310-b7d9-9358b9ac24b2
180 lines
8.9 KiB
XML
180 lines
8.9 KiB
XML
<!--
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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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<profile name="internal">
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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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<alias name="default"/>
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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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<param name="sip-trace" value="no"/>
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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="100"/>
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<param name="codec-prefs" value="$${global_codec_prefs}"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
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<param name="rtp-ip" value="$${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="rfc1918"/>-->
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<!--<param name="aggressive-nat-detection" value="true"/>-->
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<!--<param name="enable-timer" value="false"/>-->
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<!--<param name="enable-100rel" 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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<!--<param name="apply-register-acl" value="domains"/>-->
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<!--<param name="dtmf-type" value="info"/>-->
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<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
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<!--enable to use presence and mwi -->
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<param name="manage-presence" value="true"/>
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<!-- used to share presence info across sofia profiles -->
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<!-- Name of the db to use for this profile -->
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<!--<param name="dbname" value="share_presence"/>-->
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<!--<param name="presence-hosts" value="$${domain}"/>-->
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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="120"/>-->
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<!--<param name="multiple-registrations" value="true"/>-->
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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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<!--If you don't want to pass through timestampes 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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<!-- 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="$${internal_auth_calls}"/>
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<!-- on authed calls, authenticate *all* the packets not just invite -->
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<param name="auth-all-packets" value="false"/>
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<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
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<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
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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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<!--all inbound reg will look in this domain for the users -->
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<!--<param name="force-register-domain" value="cluecon.com"/>-->
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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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<!-- enable-3pcc can be set to either 'true' or 'proxy', true accepts the call right away, proxy waits until the call has been answered then sends accepts -->
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<!--<param name="enable-3pcc" value="true"/>-->
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<!-- use stun when specified (default is true) -->
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<!--<param name="stun-enabled" value="true"/>-->
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<!-- use stun when specified (default is true) -->
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<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
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<!--<param name="stun-auto-disable" 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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</settings>
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</profile>
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