net/freeswitch/files/etc.uci/sip_profiles/internal.xml

   1 <profile name="internal">
   2   <!--
   3       This is a sofia sip profile/user agent.  This will service exactly one ip and port.
   4       In FreeSWITCH you can run multiple sip user agents on their own ip and port.
   5
   6       When you hear someone say "sofia profile" this is what they are talking about.
   7   -->
   8
   9   <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  10   <!--aliases are other names that will work as a valid profile name for this profile-->
  11   <aliases>
  12     <!--
  13     <alias name="default"/>
  14     -->
  15   </aliases>
  16   <!-- Outbound Registrations -->
  17   <gateways>
  18     <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  19   </gateways>
  20
  21   <domains>
  22     <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
  23     <!--<domain name="$${domain}" parse="true"/>-->
  24     <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
  25     <!--<domain name="all" alias="true" parse="true"/>-->
  26     <domain name="all" alias="true" parse="false"/>
  27   </domains>
  28
  29   <settings>
  30     <!--
  31         When calls are in no media this will bring them back to media
  32         when you press the hold button.
  33     -->
  34     <!--<param name="media-option" value="resume-media-on-hold"/> -->
  35     <!--
  36         This will allow a call after an attended transfer go back to
  37          bypass media after an attended transfer.
  38     -->
  39     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
  40     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
  41     <param name="debug" value="0"/>
  42         <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
  43         <!-- <param name="shutdown-on-fail" value="true"/> -->
  44     <param name="sip-trace" value="no"/>
  45     <param name="log-auth-failures" value="true"/>
  46     <param name="context" value="public"/>
  47     <param name="rfc2833-pt" value="101"/>
  48     <!-- port to bind to for sip traffic -->
  49     <param name="sip-port" value="$${internal_sip_port}"/>
  50     <param name="dialplan" value="XML"/>
  51     <param name="dtmf-duration" value="2000"/>
  52     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
  53     <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
  54     <param name="rtp-timer-name" value="soft"/>
  55     <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
  56     <param name="rtp-ip" value="$${local_ip_v4}"/>
  57     <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
  58     <param name="sip-ip" value="$${local_ip_v4}"/>
  59     <param name="hold-music" value="$${hold_music}"/>
  60     <param name="apply-nat-acl" value="nat.auto"/>
  61
  62     <!-- extended info parsing -->
  63     <!-- <param name="extended-info-parsing" value="true"/> -->
  64
  65     <!--<param name="aggressive-nat-detection" value="true"/>-->
  66     <!--
  67         There are known issues (asserts and segfaults) when 100rel is enabled.
  68         It is not recommended to enable 100rel at this time.
  69     -->
  70     <!--<param name="enable-100rel" value="true"/>-->
  71     <!-- Enable Compact SIP headers. -->
  72     <!--<param name="enable-compact-headers" value="true"/>-->
  73     <!--
  74         enable/disable session timers
  75     -->
  76     <!--<param name="enable-timer" value="false"/>-->
  77     <!--<param name="minimum-session-expires" value="120"/>-->
  78     <param name="apply-inbound-acl" value="domains"/>
  79     <!--
  80         This defines your local network, by default we detect your local network
  81         and create this localnet.auto ACL for this.
  82     -->
  83     <param name="local-network-acl" value="localnet.auto"/>
  84     <!--<param name="apply-register-acl" value="domains"/>-->
  85     <!--<param name="dtmf-type" value="info"/>-->
  86
  87
  88     <!-- 'true' means every time 'first-only' means on the first register -->
  89     <!--<param name="send-message-query-on-register" value="true"/>-->
  90
  91
  92
  93     <param name="record-path" value="$${recordings_dir}"/>
  94     <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
  95     <!--enable to use presence -->
  96     <param name="manage-presence" value="true"/>
  97     <!--<param name="manage-shared-appearance" value="true"/>-->
  98     <!-- used to share presence info across sofia profiles -->
  99     <!-- Name of the db to use for this profile -->
 100     <!--<param name="dbname" value="share_presence"/>-->
 101     <!--<param name="presence-hosts" value="$${domain}"/>-->
 102     <!-- ************************************************* -->
 103
 104     <!-- This setting is for AAL2 bitpacking on G726 -->
 105     <!-- <param name="bitpacking" value="aal2"/> -->
 106     <!--max number of open dialogs in proceeding -->
 107     <!--<param name="max-proceeding" value="1000"/>-->
 108     <!--session timers for all call to expire after the specified seconds -->
 109     <!--<param name="session-timeout" value="120"/>-->
 110     <!-- Can be 'true' or 'contact' -->
 111     <!--<param name="multiple-registrations" value="contact"/>-->
 112     <!--set to 'greedy' if you want your codec list to take precedence -->
 113     <param name="inbound-codec-negotiation" value="generous"/>
 114     <!-- if you want to send any special bind params of your own -->
 115     <!--<param name="bind-params" value="transport=udp"/>-->
 116     <!--<param name="unregister-on-options-fail" value="true"/>-->
 117
 118     <!-- TLS: disabled by default, set to "true" to enable -->
 119     <param name="tls" value="$${internal_ssl_enable}"/>
 120     <!-- additional bind parameters for TLS -->
 121     <param name="tls-bind-params" value="transport=tls"/>
 122     <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
 123     <param name="tls-sip-port" value="$${internal_tls_port}"/>
 124     <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
 125     <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
 126     <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
 127     <param name="tls-version" value="$${sip_tls_version}"/>
 128
 129     <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
 130          (reduces delay on latent connections default true, must be disabled explicitly)-->
 131     <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
 132
 133     <!--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)-->
 134     <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
 135     <!--<param name="pass-rfc2833" value="true"/>-->
 136     <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
 137     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
 138
 139     <!--Uncomment to set all inbound calls to no media mode-->
 140     <!--<param name="inbound-bypass-media" value="true"/>-->
 141
 142     <!--Uncomment to set all inbound calls to proxy media mode-->
 143     <!--<param name="inbound-proxy-media" value="true"/>-->
 144
 145     <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
 146     <!--<param name="inbound-late-negotiation" value="true"/>-->
 147
 148     <!-- this lets anything register -->
 149     <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
 150     <!-- <param name="accept-blind-reg" value="true"/> -->
 151
 152     <!-- accept any authentication without actually checking (not a good feature for most people) -->
 153     <!-- <param name="accept-blind-auth" value="true"/> -->
 154
 155     <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
 156     <!-- <param name="suppress-cng" value="true"/> -->
 157
 158     <!--TTL for nonce in sip auth-->
 159     <param name="nonce-ttl" value="60"/>
 160     <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
 161         that the originator is using-->
 162     <!--<param name="disable-transcoding" value="true"/>-->
 163     <!-- Handle 302 Redirect in the dialplan -->
 164     <!--<param name="manual-redirect" value="true"/> -->
 165     <!-- Disable Transfer -->
 166     <!--<param name="disable-transfer" value="true"/> -->
 167     <!-- Disable Register -->
 168     <!--<param name="disable-register" value="true"/> -->
 169     <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
 170     <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
 171     <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
 172     <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
 173     <param name="auth-calls" value="$${internal_auth_calls}"/>
 174     <!-- Force the user and auth-user to match. -->
 175     <param name="inbound-reg-force-matching-username" value="true"/>
 176     <!-- on authed calls, authenticate *all* the packets not just invite -->
 177     <param name="auth-all-packets" value="false"/>
 178
 179     <!-- external_sip_ip
 180       Used as the public IP address for SDP.
 181       Can be an one of:
 182            ip address            - "12.34.56.78"
 183            a stun server lookup  - "stun:stun.server.com"
 184            a DNS name            - "host:host.server.com"
 185            auto                  - Use guessed ip.
 186            auto-nat              - Use ip learned from NAT-PMP or UPNP
 187        -->
 188     <param name="ext-rtp-ip" value="auto-nat"/>
 189     <param name="ext-sip-ip" value="auto-nat"/>
 190
 191     <!-- rtp inactivity timeout -->
 192     <param name="rtp-timeout-sec" value="300"/>
 193     <param name="rtp-hold-timeout-sec" value="1800"/>
 194     <!-- VAD choose one (out is a good choice); -->
 195     <!-- <param name="vad" value="in"/> -->
 196     <!-- <param name="vad" value="out"/> -->
 197     <!-- <param name="vad" value="both"/> -->
 198     <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
 199     <!--
 200         These are enabled to make the default config work better out of the box.
 201         If you need more than ONE domain you'll need to not use these options.
 202
 203     -->
 204     <!--all inbound reg will look in this domain for the users -->
 205     <param name="force-register-domain" value="$${domain}"/>
 206     <!--force the domain in subscriptions to this value -->
 207     <param name="force-subscription-domain" value="$${domain}"/>
 208     <!--all inbound reg will stored in the db using this domain -->
 209     <param name="force-register-db-domain" value="$${domain}"/>
 210     <!--force suscription expires to a lower value than requested-->
 211     <!--<param name="force-subscription-expires" value="60"/>-->
 212     <!-- disable register and transfer which may be undesirable in a public switch -->
 213     <!--<param name="disable-transfer" value="true"/>-->
 214     <!--<param name="disable-register" value="true"/>-->
 215
 216     <!--
 217          enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
 218          right away, proxy waits until the call has been answered then sends accepts
 219     -->
 220     <!--<param name="enable-3pcc" value="true"/>-->
 221
 222     <!-- use at your own risk or if you know what this does.-->
 223     <!--<param name="NDLB-force-rport" value="true"/>-->
 224     <!--
 225         Choose the realm challenge key. Default is auto_to if not set.
 226
 227         auto_from  - uses the from field as the value for the sip realm.
 228         auto_to    - uses the to field as the value for the sip realm.
 229         <anyvalue> - you can input any value to use for the sip realm.
 230
 231         If you want URL dialing to work you'll want to set this to auto_from.
 232
 233         If you use any other value besides auto_to or auto_from you'll loose
 234         the ability to do multiple domains.
 235
 236         Note: comment out to restore the behavior before 2008-09-29
 237
 238     -->
 239     <param name="challenge-realm" value="auto_from"/>
 240     <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
 241     <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
 242     <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
 243     <!-- on outbound calls set the callid to match the uuid of the session -->
 244     <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
 245     <!-- set to false disable this feature -->
 246     <!--<param name="rtp-autofix-timing" value="false"/>-->
 247
 248     <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
 249     <!--<param name="pass-callee-id" value="false"/>-->
 250
 251     <!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
 252          valid values:
 253
 254          clear
 255          CISCO_SKIP_MARK_BIT_2833
 256          SONUS_SEND_INVALID_TIMESTAMP_2833
 257
 258     -->
 259     <!--<param name="auto-rtp-bugs" data="clear"/>-->
 260
 261          <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
 262          <!--<param name="disable-srv" value="false" />-->
 263          <!--<param name="disable-naptr" value="false" />-->
 264
 265         <!-- The following can be used to fine-tune timers within sofia's transport layer
 266                  Those settings are for advanced users and can safely be left as-is -->
 267
 268         <!-- Initial retransmission interval (in milliseconds).
 269                 Set the T1 retransmission interval used by the SIP transaction engine.
 270                 The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G.   -->
 271         <!-- <param name="timer-T1" value="500" /> -->
 272
 273         <!--  Transaction timeout (defaults to T1 * 64).
 274                 Set the T1x64 timeout value used by the SIP transaction engine.
 275                 The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
 276                 The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
 277         <!-- <param name="timer-T1X64" value="32000" /> -->
 278
 279
 280         <!-- Maximum retransmission interval (in milliseconds).
 281                 Set the maximum retransmission interval used by the SIP transaction engine.
 282                 The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
 283                 Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
 284                 until the timer B fires.  -->
 285         <!-- <param name="timer-T2" value="4000" /> -->
 286
 287         <!--
 288                 Transaction lifetime (in milliseconds).
 289                 Set the lifetime for completed transactions used by the SIP transaction engine.
 290                 A completed transaction is kept around for the duration of T4 in order to catch late responses.
 291                 The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
 292         <!-- <param name="timer-T4" value="4000" /> -->
 293
 294   </settings>
 295 </profile>
 296