<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal" domain="$${domain}"> <!--aliases are other names that will work as a valid profile name for this profile--> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <!-- Outbound Registrations --> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> <!-- indicator to parse the directory for domains with parse="true" to get gateways--> <!--<domain name="$${domain}" parse="true"/>--> <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile --> <!--<domain name="all" alias="true" parse="true"/>--> </domains> <settings> <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> --> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <!-- port to bind to for sip traffic --> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <!-- ip address to use for rtp --> <param name="rtp-ip" value="$${local_ip_v4}"/> <!-- ip address to bind to --> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <!--<param name="apply-nat-acl" value="rfc1918"/>--> <!--<param name="aggressive-nat-detection" value="true"/>--> <!--<param name="enable-timer" value="false"/>--> <!--<param name="enable-100rel" value="false"/>--> <param name="apply-inbound-acl" value="domains"/> <!--<param name="apply-register-acl" value="domains"/>--> <!--<param name="dtmf-type" value="info"/>--> <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <!--enable to use presence and mwi --> <param name="manage-presence" value="true"/> <!-- used to share presence info across sofia profiles --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!-- ************************************************* --> <!-- This setting is for AAL2 bitpacking on G726 --> <!-- <param name="bitpacking" value="aal2"/> --> <!--max number of open dialogs in proceeding --> <!--<param name="max-proceeding" value="1000"/>--> <!--session timers for all call to expire after the specified seconds --> <!--<param name="session-timeout" value="120"/>--> <!--<param name="multiple-registrations" value="true"/>--> <!--set to 'greedy' if you want your codec list to take precedence --> <param name="inbound-codec-negotiation" value="generous"/> <!-- if you want to send any special bind params of your own --> <!--<param name="bind-params" value="transport=udp"/>--> <!--<param name="unregister-on-options-fail" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) --> <param name="tls-sip-port" value="5061"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> <param name="tls-version" value="tlsv1"/> <!--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)--> <!--<param name="rtp-rewrite-timestamps" value="true"/>--> <!--<param name="pass-rfc2833" value="true"/>--> <!--If you have ODBC support and a working dsn you can use it instead of SQLite--> <!--<param name="odbc-dsn" value="dsn:user:pass"/>--> <!--Uncomment to set all inbound calls to no media mode--> <!--<param name="inbound-bypass-media" value="true"/>--> <!--Uncomment to set all inbound calls to proxy media mode--> <!--<param name="inbound-proxy-media" value="true"/>--> <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <!--<param name="inbound-late-negotiation" value="true"/>--> <!-- this lets anything register --> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <!-- <param name="accept-blind-reg" value="true"/> --> <!-- accept any authentication without actually checking (not a good feature for most people) --> <!-- <param name="accept-blind-auth" value="true"/> --> <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable --> <!-- <param name="suppress-cng" value="true"/> --> <!--TTL for nonce in sip auth--> <param name="nonce-ttl" value="60"/> <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec that the originator is using--> <!--<param name="disable-transcoding" value="true"/>--> <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash --> <!--<param name="NDLB-broken-auth-hash" value="true"/>--> <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling --> <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>--> <param name="auth-calls" value="true"/> <!-- on authed calls, authenticate *all* the packets not just invite --> <param name="auth-all-packets" value="false"/> <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <!-- rtp inactivity timeout --> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:10.0.1.251:5555"/>--> <!--all inbound reg will look in this domain for the users --> <!--<param name="force-register-domain" value="cluecon.com"/>--> <!-- disable register and transfer which may be undesirable in a public switch --> <!--<param name="disable-transfer" value="true"/>--> <!--<param name="disable-register" value="true"/>--> <!--<param name="enable-3pcc" value="true"/>--> <!-- use stun when specified (default is true) --> <!--<param name="stun-enabled" value="true"/>--> <!-- use stun when specified (default is true) --> <!-- set to true to have the profile determine stun is not useful and turn it off globally--> <!--<param name="stun-auto-disable" value="true"/>--> </settings> </profile>