conf/insideout/sip_profiles/internal.xml

   1 <!--
   2     This is a sofia sip profile/user agent.  This will service exactly one ip and port.
   3     In FreeSWITCH you can run multiple sip user agents on their own ip and port.
   4
   5     When you hear someone say "sofia profile" this is what they are talking about.
   6 -->
   7
   8 <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
   9 <profile name="internal">
  10   <!--aliases are other names that will work as a valid profile name for this profile-->
  11   <aliases>
  12     <alias name="local"/>
  13   </aliases>
  14
  15   <!-- Outbound Registrations -->
  16   <gateways>
  17     <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  18   </gateways>
  19
  20   <domains>
  21     <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
  22     <!--<domain name="$${domain}" parse="true"/>-->
  23     <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
  24     <!--<domain name="all" alias="true" parse="true"/>-->
  25     <domain name="all" alias="true" parse="false"/>
  26   </domains>
  27
  28   <settings>
  29     <!--
  30         When calls are in no media this will bring them back to media
  31         when you press the hold button.
  32     -->
  33     <!--<param name="media-option" value="resume-media-on-hold"/> -->
  34     <!--
  35         This will allow a call after an attended transfer go back to
  36         bypass media after an attended transfer.
  37     -->
  38     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
  39
  40     <!-- Can be set to "_undef_" to remove the User-Agent header -->
  41     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
  42     <param name="debug" value="0"/>
  43     <param name="sip-trace" value="no"/>
  44     <param name="context" value="public"/>
  45     <param name="rfc2833-pt" value="101"/>
  46     <!-- port to bind to for sip traffic -->
  47     <param name="sip-port" value="$${internal_sip_port}"/>
  48     <param name="dialplan" value="enum,XML,lcr"/>
  49     <param name="dtmf-duration" value="100"/>
  50     <param name="codec-prefs" value="$${global_codec_prefs}"/>
  51     <param name="use-rtp-timer" value="true"/>
  52     <param name="rtp-timer-name" value="soft"/>
  53     <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
  54     <param name="rtp-ip" value="$${internal_ip_v4}"/>
  55     <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
  56     <param name="sip-ip" value="$${internal_ip_v4}"/>
  57     <param name="hold-music" value="$${hold_music}"/>
  58     <!--<param name="apply-nat-acl" value="rfc1918"/>-->
  59     <!--<param name="aggressive-nat-detection" value="true"/>-->
  60     <!--<param name="enable-timer" value="false"/>-->
  61     <!--<param name="enable-100rel" value="true"/>-->
  62     <!--<param name="minimum-session-expires" value="120"/>-->
  63     <param name="apply-inbound-acl" value="domains"/>
  64     <!--<param name="apply-register-acl" value="domains"/>-->
  65     <!--<param name="dtmf-type" value="info"/>-->
  66     <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
  67
  68
  69     <!-- This setting is for AAL2 bitpacking on G726 -->
  70     <!-- <param name="bitpacking" value="aal2"/> -->
  71     <!--max number of open dialogs in proceeding -->
  72     <!--<param name="max-proceeding" value="1000"/>-->
  73     <!--session timers for all call to expire after the specified seconds -->
  74     <!--<param name="session-timeout" value="120"/>-->
  75     <!-- Can be 'true' or 'contact' -->
  76     <!--<param name="multiple-registrations" value="contact"/>-->
  77     <!--set to 'greedy' if you want your codec list to take precedence -->
  78     <param name="inbound-codec-negotiation" value="generous"/>
  79     <!-- if you want to send any special bind params of your own -->
  80     <!--<param name="bind-params" value="transport=udp"/>-->
  81     <!--<param name="unregister-on-options-fail" value="true"/>-->
  82
  83     <!-- TLS: disabled by default, set to "true" to enable -->
  84     <param name="tls" value="$${internal_ssl_enable}"/>
  85     <!-- additional bind parameters for TLS -->
  86     <param name="tls-bind-params" value="transport=tls"/>
  87     <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
  88     <param name="tls-sip-port" value="$${internal_tls_port}"/>
  89     <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
  90     <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
  91     <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
  92     <param name="tls-version" value="$${sip_tls_version}"/>
  93
  94     <!--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)-->
  95     <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
  96     <!--<param name="pass-rfc2833" value="true"/>-->
  97     <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
  98     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
  99
 100     <!--Uncomment to set all inbound calls to no media mode-->
 101     <!--<param name="inbound-bypass-media" value="true"/>-->
 102
 103     <!--Uncomment to set all inbound calls to proxy media mode-->
 104     <!--<param name="inbound-proxy-media" value="true"/>-->
 105
 106     <!-- Let calls hit the dialplan before selecting codec for the a-leg -->
 107     <param name="inbound-late-negotiation" value="true"/>
 108
 109     <!-- this lets anything register -->
 110     <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
 111     <!-- <param name="accept-blind-reg" value="true"/> -->
 112
 113     <!-- accept any authentication without actually checking (not a good feature for most people) -->
 114     <!-- <param name="accept-blind-auth" value="true"/> -->
 115
 116     <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
 117     <!-- <param name="suppress-cng" value="true"/> -->
 118
 119     <!--TTL for nonce in sip auth-->
 120     <param name="nonce-ttl" value="60"/>
 121     <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
 122         that the originator is using-->
 123     <!--<param name="disable-transcoding" value="true"/>-->
 124     <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
 125     <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
 126     <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
 127     <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
 128     <param name="auth-calls" value="$${internal_auth_calls}"/>
 129     <!-- Force subscription requests to require authentication -->
 130     <param name="auth-subscriptions" value="true"/>
 131     <!-- Force the user and auth-user to match. -->
 132     <param name="inbound-reg-force-matching-username" value="true"/>
 133     <!-- on authed calls, authenticate *all* the packets not just invite -->
 134     <param name="auth-all-packets" value="false"/>
 135     <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
 136     <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
 137     <!-- rtp inactivity timeout -->
 138     <param name="rtp-timeout-sec" value="300"/>
 139     <param name="rtp-hold-timeout-sec" value="1800"/>
 140     <!-- VAD choose one (out is a good choice); -->
 141     <!-- <param name="vad" value="in"/> -->
 142     <!-- <param name="vad" value="out"/> -->
 143     <!-- <param name="vad" value="both"/> -->
 144     <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
 145
 146     <!--all inbound reg will look in this domain for the users -->
 147     <param name="force-register-domain" value="$${domain}"/>
 148     <!--all inbound reg will stored in the db using this domain -->
 149     <param name="force-register-db-domain" value="$${domain}"/>
 150     <!--enable to use presence -->
 151     <param name="manage-presence" value="true"/>
 152     <!-- used to share presence info across sofia profiles -->
 153     <!-- Name of the db to use for this profile -->
 154     <param name="dbname" value="$${domain}"/>
 155     <param name="presence-hosts" value="$${domain}"/>
 156     <!-- ************************************************* -->
 157
 158
 159     <!--force suscription expires to a lower value than requested-->
 160     <!--<param name="force-subscription-expires" value="60"/>-->
 161     <!-- disable register and transfer which may be undesirable in a public switch -->
 162     <!--<param name="disable-transfer" value="true"/>-->
 163     <!--<param name="disable-register" value="true"/>-->
 164
 165     <!-- 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 -->
 166     <!--<param name="enable-3pcc" value="true"/>-->
 167
 168     <!-- use at your own risk or if you know what this does.-->
 169     <!--<param name="NDLB-force-rport" value="true"/>-->
 170     <!--
 171         Choose the realm challenge key. Default is auto_to if not set.
 172
 173         auto_from  - uses the from field as the value for the sip realm.
 174         auto_to    - uses the to field as the value for the sip realm.
 175         <anyvalue> - you can input any value to use for the sip realm.
 176
 177         If you want URL dialing to work you'll want to set this to auto_from.
 178
 179         If you use any other value besides auto_to or auto_from you'll loose
 180         the ability to do multiple domains.
 181
 182         Note: comment out to restore the behavior before 2008-09-29
 183
 184     -->
 185     <param name="challenge-realm" value="auto_from"/>
 186     <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
 187     <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
 188     <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
 189     <!-- on outbound calls set the callid to match the uuid of the session -->
 190     <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
 191   </settings>
 192 </profile>