[Freeswitch-users] Unable to Configure FreeSwitch

7 Jun 2007

      I am using freeswitch on windows platform and trying to configure it with
gafachi account. 

I am trying to make outbound calls but the wireshark traces shows the From
address given below: 

From: "102" <sip:103@192.168.96.57>; tag=D5NmFsgvr8ej

When I dial the number with xlite , I am able to connect to other phone. 

Will any body tell me what wrong I did? 

Here are my configuration details in freeswithc.conf.

<?xml version="1.0"?>

<document type="freeswitch/xml">

  #set "domain=abc"

  #set "subdomain=192.168.96.57"

  <!--#set "default_codecs=PCMU@20i"-->

  <!--my domain is $${domain}-->

  <section name="configuration" description="Various Configuration">

    <configuration name="switch.conf" description="Modules">

      <settings>

      <!--Most channels to allow at once -->

      <param name="max-sessions" value="1000"/>

      </settings>

      <!--Any variables defined here will be available in every channel, in
the dialplan etc -->

      <variables>

      <variable name="uk-ring"
value="%(400,200,400,450);%(400,2200,400,450)"/>

      <variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/>

      <variable name="bong-ring"
value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/>

      </variables>

    </configuration>

    <configuration name="modules.conf" description="Modules">

      <modules>

      <!-- Loggers (I'd load these first) -->

       <load module="mod_console"/> 

       <load module="mod_syslog"/> 

      <!-- Multi-Faceted -->

      <!-- mod_enum is a dialplan interface, an application interface and an
api command interface -->

      <load module="mod_enum"/>

      <!-- XML Interfaces -->

      <!-- <load module="mod_xml_rpc"/> -->

      <!-- <load module="mod_xml_curl"/> -->

      <!-- Event Handlers -->

       <load module="mod_cdr"/> 

      <!-- <load module="mod_event_multicast"/> -->

      <load module="mod_event_socket"/> 

      <!-- <load module="mod_xmpp_event"/> -->

      <!-- <load module="mod_zeroconf"/> -->

      <!-- Directory Interfaces -->

      <!-- <load module="mod_ldap"/> -->

      <!-- Endpoints -->

      <!-- <load module="mod_dingaling"/> -->

      <!--<load module="mod_iax"/>-->

      <load module="mod_portaudio"/>

      <load module="mod_sofia"/>

      <!-- <load module="mod_wanpipe"/> -->

      <!-- <load module="mod_woomera"/> -->

      <!-- Applications -->

      <load module="mod_bridgecall"/>

      <load module="mod_commands"/>

      <load module="mod_conference"/>

      <load module="mod_dptools"/>

      <load module="mod_echo"/>

      <!--<load module="mod_park"/>-->

      <load module="mod_playback"/>

      <!-- Dialplan Interfaces -->

      <!-- <load module="mod_dialplan_directory"/> -->

      <load module="mod_dialplan_xml"/>

      <!-- Codec Interfaces -->

      <load module="mod_g711"/>

      <load module="mod_gsm"/>

      <!-- <load module="mod_ilbc"/> -->

      <load module="mod_l16"/>

      <!-- <load module="mod_speex"/> -->

      <!-- File Format Interfaces -->

      <load module="mod_sndfile"/>

      <load module="mod_native_file"/>

    <!--For icecast/mp3 streams/files-->

    <!--<load module="mod_shout"/>-->

      <!-- Timers -->

      <load module="mod_softtimer"/>

      <!-- Languages -->

       <load module="mod_spidermonkey"/> 

      <!-- <load module="mod_perl"/> -->

      <!-- ASR /TTS -->

      <load module="mod_cepstral"/> 

      <!-- <load module="mod_rss"/> -->

    <!-- Say -->

    <load module="mod_say_en"/>

      </modules>

    </configuration>

    <configuration name="console.conf" description="Console Logger">

      <!-- pick a file name, a function name or 'all' -->

      <!-- map as many as you need for specific debugging -->

      <mappings>

      <!-- <param name="log_event" value="DEBUG"/> -->

      <param name="all" value="DEBUG"/>

      </mappings>

    </configuration>

    <configuration name="sofia.conf" description="sofia Endpoint">

      <profiles>

      <!-- <profile name="mydomain1.com"> -->

      <profile name="$${domain}">

            <!--<registrations>

        <registration name="16462781042">

             <param name="register-scheme" value="friend"/>

             <param name="register-realm" value="gafachi"/>

             <param name="register-username" value="username"/>

             <param name="register-password" value="password"/>

             <param name="register-from" value="sip:username@gafachi"/>

             <param name="register-to" value="sip:username@gafachi"/>

             <param name="register-proxy"
value="username.sip.gafachi.com:5060"/>

             <param name="register-frequency" value="20"/>

             </registration> 

        </registrations>-->

    <gateways>

      <gateway name="192.168.96.57">

        <!---->/// account username *required* ///-->

        <param name="username" value="username"/>

        /// auth realm: *optional* same as gateway name, if blank ///

        <param name="realm" value="domain"/>

        /// account password *required* ///

        <param name="password" value="password"/>

        /// extension for inbound calls: *optional* same as username, if
blank ///

        <!--<param name="extension" value="username"/>-->

        /// proxy host: *optional* same as realm, if blank ///

        <param name="proxy" value="realm"/>

        /// expire in seconds: *optional* 3600, if blank ///

        <param name="expire-seconds" value="60"/>

      </gateway>

    </gateways> 

        <settings>

          <param name="debug" value="1"/>

          <param name="rfc2833-pt" value="101"/>

          <param name="sip-port" value="5060"/>

          <param name="dialplan" value="XML"/>

          <param name="dtmf-duration" value="100"/>

          <param name="codec-prefs" value="PCMU@20i"/>

          <param name="codec-ms" value="20"/>

          <param name="use-rtp-timer" value="true"/>

          <param name="rtp-timer-name" value="soft"/>

          <param name="rtp-ip" value="192.168.96.57"/>

          <param name="sip-ip" value="192.168.96.57"/>

          <!--Uncomment to set all inbound calls to no media mode-->

          <!--<param name="inbound-no-media" 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"/>

          <!--<param name="auth-calls" value="true"/>-->

          <!-- on authed calls, authenticate *all* the packets not just
invite -->

          <!--<param name="auth-all-packets" value="true"/>-->

          <!-- optional ; -->

          <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->

          <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->

          <!-- 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"/>-->

        </settings>

      </profile>

      </profiles>

    </configuration>

Thanks

Dinesh

[Freeswitch-users] Unable to Configure FreeSwitch

Dinesh Dialani