FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.<br>
<br><br>freeswitch.xml:<br>
<section name="directory" description="User Directory"><br>
<X-PRE-PROCESS cmd="include" data="directory/*.xml"/><br>
<br><br><br>Than, you need to check sip profiles. By default FS will accept registrations on internal profiles only... so you should enable it on the external as well.<br><br><br>look at this portion of your adequate sip profile:<br>
<br> <!-- this lets anything register --><br> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --><br> <!-- <param name="accept-blind-reg" value="true"/> --><br>
<br> <!-- accept any authentication without actually checking (not a good feature for most people) --><br> <!-- <param name="accept-blind-auth" value="true"/> --><br><br> <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable --><br>
<!-- <param name="suppress-cng" value="true"/> --><br><br> <!--TTL for nonce in sip auth--><br> <param name="nonce-ttl" value="60"/><br> <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec <br>
that the originator is using--><br> <!--<param name="disable-transcoding" value="true"/>--><br> <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash --><br>
<!--<param name="NDLB-broken-auth-hash" value="true"/>--><br> <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling --><br>
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>--><br> <param name="auth-calls" value="$${internal_auth_calls}"/><br> <!-- Force the user and auth-user to match. --><br>
<param name="inbound-reg-force-matching-username" value="true"/><br> <!-- on authed calls, authenticate *all* the packets not just invite --><br> <param name="auth-all-packets" value="false"/><br>
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --><br> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --><br> <!-- rtp inactivity timeout --><br>
<param name="rtp-timeout-sec" value="300"/><br> <param name="rtp-hold-timeout-sec" value="1800"/><br> <!-- VAD choose one (out is a good choice); --><br>
<!-- <param name="vad" value="in"/> --><br> <!-- <param name="vad" value="out"/> --><br> <!-- <param name="vad" value="both"/> --><br>
<!--<param name="alias" value="sip:<a href="http://10.0.1.251:5555">10.0.1.251:5555</a>"/>--><br> <!--<br> These are enabled to make the default config work better out of the box.<br>
If you need more than ONE domain you'll need to not use these options.<br><br> --><br> <!--all inbound reg will look in this domain for the users --><br> <param name="force-register-domain" value="$${domain}"/><br>
<!--all inbound reg will stored in the db using this domain --><br> <param name="force-register-db-domain" value="$${domain}"/><br> <!--force suscription expires to a lower value than requested--><br>
<!--<param name="force-subscription-expires" value="60"/>--><br> <!-- disable register and transfer which may be undesirable in a public switch --><br> <!--<param name="disable-transfer" value="true"/>--><br>
<!--<param name="disable-register" value="true"/>--><br><br><br><br>Just make sure you use correct IP_ADDRESS:PORT to match the correct profile<br>
<br>vars.xml:<br><br> <!-- Internal SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/><br> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/><br>
<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/><br><br><br> <!-- External SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/><br>
<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/><br> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/><br><br><br>T.<br><br><br><div class="gmail_quote">
On Wed, Sep 16, 2009 at 11:29 AM, pankaj anand <span dir="ltr"><<a href="mailto:pankajanand18@gmail.com" target="_blank">pankajanand18@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
hi , <div> i m very new to the FreeSwitch..</div><div>can any one tell me how to add a new user. </div><div>i have already tried creating a new user by creating a $INSTALL_DIR/conf/directory/default/pankaj.xml :</div><div>
<br></div><div><div><include></div><div> <user id="pankaj"></div><div> <params></div><div> <param name="password" value="pankaj"/></div><div> <param name="vm-password" value="pankaj"/></div>
<div> </params></div><div> <variables></div><div> <variable name="toll_allow" value="domestic,international,local"/></div><div> <variable name="accountcode" value="pankaj"/></div>
<div> <variable name="user_context" value="default"/></div><div> <variable name="effective_caller_id_name" value="Extension pankaj"/></div><div> <variable name="effective_caller_id_number" value="pankaj"/></div>
<div> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/></div><div> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/></div>
<div> <variable name="callgroup" value="techsupport"/></div><div> </variables></div><div> </user></div><div></include></div><div><br></div><div>but when i try to connect it using , the softphone shows forbidden.</div>
<div>Can anyone tell me where i am making a mistake.</div><div><br></div><div>with regards</div><div>Pankaj anand</div><div><br></div></div><div><br></div><div><br></div>
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