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Comment: sip_secure_media => rtp_secure_media

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Introduction

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Code Block
<include>
  <!--
      ipauth if you have an cidr= in the user attributes ie cidr="1.2.3.4/32"  
      see <node type="allow" domain="$${domain}"/> in default acl.conf.xml 
  -->
  <user id="brian" cidr="192.0.2.0/24">
    <!-- Outbound Registrations Related to this user -->
    <gateways>
      <!--<gateway name="asterlink.com">-->
      <!--/// account username *required* ///-->
      <!--<param name="username" value="cluecon"/>-->
      <!--/// auth realm: *optional* same as gateway name, if blank ///-->
      <!--<param name="realm" value="asterlink.com"/>-->
      <!--/// username to use in from: *optional* same as  username, if blank ///-->
      <!--<param name="from-user" value="cluecon"/>-->
      <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
      <!--<param name="from-domain" value="asterlink.com"/>-->
      <!--/// account password *required* ///-->
      <!--<param name="password" value="2007"/>--> 
      <!--/// replace the INVITE from user with the channel's caller-id ///-->
      <!--<param name="caller-id-in-from" value="false"/>-->
      <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
      <!--<param name="extension" value="cluecon"/>-->
      <!--/// proxy host: *optional* same as realm, if blank ///-->
      <!--<param name="proxy" value="asterlink.com"/>-->
      <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
      <!--<param name="register-proxy" value="mysbc.com"/>-->
      <!--/// expire in seconds: *optional* 3600, if blank ///-->
      <!--<param name="expire-seconds" value="60"/>-->
      <!--/// do not register ///-->
      <!--<param name="register" value="false"/>-->
      <!-- which transport to use for register -->
      <!--<param name="register-transport" value="udp"/>-->
      <!--How many seconds before a retry when a failure or timeout occurs -->
      <!--<param name="retry-seconds" value="30"/>-->
      <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
      <!--<param name="caller-id-in-from" value="false"/>-->
      <!--extra sip params to send in the contact-->
      <!--<param name="contact-params" value="tport=tcp"/>-->
      <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
      <!--<param name="ping" value="25"/>-->
      <!--</gateway>-->
    </gateways>
    <params>
      <!-- omit password for authless registration -->
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="9999"/><!--if vm-password is omitted password param is used-->
      <!--<param name="email-addr" value="me@mydomain.com"/>-->
      <!--<param name="vm-delete-file" value="true"/>-->
      <!--<param name="vm-attach-file" value="true"/>-->
      <!--<param name="vm-mailto" value="me@mydomain.com"/>-->
      <!--<param name="vm-email-all-messages" value="true"/>-->
      <!-- optionally use this instead if you want to store the hash of user:domain:pass-->
      <!--<param name="a1-hash" value="c6440e5de50b403206989679159de89a"/>-->
      <!-- What this user is allowed to acces --> 
      <!--<param name="http-allowed-api" value="jsapi,voicemail,status"/> -->
    </params>
    <variables>
      <!--all variables here will be set on all inbound calls that originate from this user -->
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Brian West"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
      <variable name="process_cdr" value="true"/>
      <!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
      <variable name="siprtp_secure_media" value="true"/>
      <!-- limit the max number of outgoing calls for this user -->
      <!--<variable name="max_calls" value="2"/>-->

      <!-- send presence information if FS is configured to do so -->
      <!--<variable name="presence_id" value="1000@$${domain}"/>-->

      <!-- set these to take advantage of a dialplan localized to this user -->
      <!--<variable name="numbering_plan" value="US"/>-->
      <!--<variable name="default_area_code" value="434"/>-->
      <!--<variable name="default_gateway" value="asterlink.com"/>-->
      <!--  
	   NDLB-connectile-dysfunction - Rewrite contact ip and port
	   NDLB-tls-connectile-dysfunction - Rewrite contact port only.
      -->
      <!--<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>-->
      <!--<variable name="sip-force-expires" value="10"/>-->
      <!--<variable name="sip-register-gateway" value="cluecon.com"/>-->
      <!-- Set the file format for a specific user -->
      <!--<variable name="vm_message_ext" value="mp3"/> -->
    </variables>

    <vcard>
      <!-- insert optional compliant vcard xml here-->
    </vcard>
  </user>
</include>

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