<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 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><extension name="oubout_pstn">
</i> <i><condition field="destination_number" expression="^tel:+(\d{12})$"></i><i>
</i> <i><action application="bridge" data="sofia/gateway/outbound/$1"/>
</i> <i></condition>
</i><i></extension></i></pre></div><div>or:<br><br></div><div><a 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"><<a href="mailto:jigar_desai12@hotmail.com" target="_blank">jigar_desai12@hotmail.com</a>></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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