<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">If this is to a registered trunk, and they include the tags we use in the registered contact, we will properly detect this, and use the to user instead of the request user. This allows you to register once for multiple DID’s. Otherwise what the provider is doing here is weird and pretty non standard, and you would need to handle it manually in the dial plan looking at the sip_to_user var.<div class=""><br class=""></div><div class="">Mike</div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Sep 7, 2016, at 11:38 AM, Allan Kristensen <<a href="mailto:ak@hejdu.dk" class="">ak@hejdu.dk</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Hello,<div class=""><br class=""></div><div class="">I've just setup a sip trunk with a new provider, however there are some issues receiving calls.</div><div class=""><br class=""></div><div class="">The INVITE looks like this:</div><div class=""><br class=""></div><div class=""><font face="monospace, monospace" class=""> INVITE <a href="http://sip:NCdWXsdSsaF2B@192.168.0.190:52284/" class="">sip:NCdWXsdSsaF2B@192.168.0.190:52284</a> SIP/2.0<br class=""></font></div><div class=""><font face="monospace, monospace" class=""> ....</font></div><div class=""><div class=""><font face="monospace, monospace" class=""> From: <<a href="sip:+XXXXXXXXXX@x.x.x.x" class="">sip:+XXXXXXXXXX@x.x.x.x</a>>;tag=Ud8ghaXg01821203</font></div><div class=""><font face="monospace, monospace" class=""> To: <<a href="sip:+YYYYYYYYYY@x.x.x.x" class="">sip:+YYYYYYYYYY@x.x.x.x</a>></font></div></div><div class=""><font face="monospace, monospace" class=""> ...</font></div><div class=""><br class=""></div><div class="">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 class=""></div><div class=""><br class=""></div><div class="">Then Freeswitch does this of cause:</div><div class=""><br class=""></div><div class=""><font face="monospace, monospace" class="">[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound<br class=""></font></div><div class=""><br class=""></div><div class="">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 class="">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 class="">Or is it my provider that is doing it wrong?<br class=""></div></div></div></blockquote></div><br class=""></div></body></html>