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    <div class="moz-cite-prefix">You can export tel-URI parameters
      before bridge:<br>
      <br>
      &lt;action application="export"
      data="nolocal:sip_invite_tel_params=foo;foo2;foo3=bar"/&gt;<br>
      <br>
      On 14-09-27 12:23 PM, Ben Langfeld wrote:<br>
    </div>
    <blockquote
cite="mid:CAAyX+KEdVbRU92u+s8DpNBbCnkyN2+kkekJHzmhJjGeQMZuTDw@mail.gmail.com"
      type="cite">
      <div dir="ltr">You have to rewrite tel URIs in your dialplan to a
        sofia path based on your choice of gateway.
        <div><br>
        </div>
        <div><a moz-do-not-send="true"
href="https://wiki.freeswitch.org/wiki/Dialplan_XML#Dialing_Through_A_Gateway_.28SIP_Provider.29">https://wiki.freeswitch.org/wiki/Dialplan_XML#Dialing_Through_A_Gateway_.28SIP_Provider.29</a><br>
        </div>
        <div><br>
        </div>
        <div>You can use something like this in the dialplan:</div>
        <div><br>
        </div>
        <div>
          <pre style="color:rgb(0,0,0)"><i>&lt;extension name="oubout_pstn"&gt;
</i>  <i>&lt;condition field="destination_number" expression="^tel:+(\d{12})$"&gt;</i><i>
</i>    <i>&lt;action application="bridge" data="sofia/gateway/outbound/$1"/&gt;
</i>  <i>&lt;/condition&gt;
</i><i>&lt;/extension&gt;</i></pre>
        </div>
        <div>or:<br>
          <br>
        </div>
        <div><a moz-do-not-send="true"
            href="https://wiki.freeswitch.org/wiki/Mod_translate">https://wiki.freeswitch.org/wiki/Mod_translate</a><br>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On 27 September 2014 06:54, joe <span
            dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:jigar_desai12@hotmail.com" target="_blank">jigar_desai12@hotmail.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">2014-09-27
            05:52:40.377783 [ERR] sofia_glue.c:983 URL Error! tel: uri's
            not<br>
            supported at this time<br>
            <br>
            <br>
            i am not able to set tel uri's in  sip_invite_req_uri=
            during originate<br>
             pstn outbound call from freeswitch.<br>
            <br>
            awaiting positive answer asap.<br>
            <br>
            tx<br>
            <br>
            <br>
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      <pre wrap="">_________________________________________________________________________
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