<div dir="ltr">It IS a trunk with multiple DID's and it IS registered with the provider.<div> <div>Setting the "extension" parameter however does not change anything. It's like the invite request is not "matched" with the gateway config and the option is therefore not applied.</div><div><div><br></div></div><div>It's a pretty simple config:</div><div><br></div><div><div><include></div><div> <gateway name="dev_tx1_13"></div><div> <param name="proxy" value="x.x.x.x"/></div><div> <param name="ping" value="10"/></div><div> <param name="register" value="true"/></div><div> <param name="extension" value="auto_to_user"/></div><div> <param name="username" value="NCdWXsdSsaF2B"/></div><div> <param name="password" value="<censored>"/></div><div> </gateway></div><div></include></div></div><div><br></div></div><div>I will contact the provider if the way they are doing it is non standard...</div><div><br></div><div>/Allan</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 7, 2016 at 5:56 PM, Victor Chukalovskiy <span dir="ltr"><<a href="mailto:victor.chukalovskiy@gmail.com" target="_blank">victor.chukalovskiy@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<p>I'd ask provider to make a change and send destination number in
SIP URI like most others do. <br>
</p>
<p>Especially if later you get different numbers from a different
provider, you might benefit from having it all standard from day
1.<br>
</p><div><div class="h5">
<pre cols="72"></pre>
<div>On 16-09-07 11:38 AM, Allan Kristensen
wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">Hello,
<div><br>
</div>
<div>I've just setup a sip trunk with a new provider, however
there are some issues receiving calls.</div>
<div><br>
</div>
<div>The INVITE looks like this:</div>
<div><br>
</div>
<div><font face="monospace, monospace"> INVITE <a href="http://sip:NCdWXsdSsaF2B@192.168.0.190:52284" target="_blank">sip:NCdWXsdSsaF2B@192.168.0.<wbr>190:52284</a>
SIP/2.0<br>
</font></div>
<div><font face="monospace, monospace"> ....</font></div>
<div>
<div><font face="monospace, monospace"> From:
<a href="mailto:sip:+XXXXXXXXXX@x.x.x.x" target="_blank"><sip:+XXXXXXXXXX@x.x.x.x></a>;tag=<wbr>Ud8ghaXg01821203</font></div>
<div><font face="monospace, monospace"> To:
<a href="mailto:sip:+YYYYYYYYYY@x.x.x.x" target="_blank"><sip:+YYYYYYYYYY@x.x.x.x></a></font></div>
</div>
<div><font face="monospace, monospace"> ...</font></div>
<div><br>
</div>
<div>The invite has my account name, instead of the destination
number and the real destination number is in the "To:" field
(which it normally is I guess ;-)<br>
</div>
<div><br>
</div>
<div>Then Freeswitch does this of cause:</div>
<div><br>
</div>
<div><font face="monospace, monospace">[INFO]
mod_dialplan_xml.c:637 Processing
+XXXXXXXXXX <+XXXXXXXXXX>-><wbr>NCdWXsdSsaF2B in context
inbound<br>
</font></div>
<div><br>
</div>
<div>That is undesirable, so Is there any way for Freeswitch to
grab the destination number from the to field instead of the
INVITE string ?</div>
<div>I could make a special sip profile for that provider with
it's own dialplan and then set destination_number from the
"sip_to_user" variable and transfer to the real dialplan, but
it just seems so messy...</div>
<div>Or is it my provider that is doing it wrong?<br>
</div>
<div><br>
</div>
<div>Best regards,</div>
<div> Allan</div>
</div>
<br>
<fieldset></fieldset>
<br>
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