<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div></div><div>I think you need to explain what you are trying to achieve.. </div><div>What you are doing currently is registering a user against freeswitch - not dissimilar to logging in to Skype. </div><div>In your bridge statement you are calling the registered user - again like you would in Skype.</div><div>Is there a reason you want asterisk to register against freeswitch? The only thing I can think of is that it's on a private LAN and that your asterisk box does something "special"..</div><div><br></div><div><br>On 2 Feb 2016, at 19:28, Marc S <<a href="mailto:therebel22@gmail.com">therebel22@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">It seems that "freeswitch gateway username" return all results about setup username and password on FS to register against external SIP gateway, am i wrong ?<br></div><div class="gmail_extra"><br><div class="gmail_quote">2016-02-02 20:23 GMT+01:00 Oz Mortimer <span dir="ltr"><<a href="mailto:omortimer@gmail.com" target="_blank">omortimer@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div></div><div>Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. </div><div>I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".</div><div>Google is your friend.. </div><div><div class="h5"><div><br></div><div><br>On 2 Feb 2016, at 19:03, Marc S <<a href="mailto:therebel22@gmail.com" target="_blank">therebel22@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">I want to use mod xml curl to generate dynamic dialplan xml like this :<br><div><br> <extension name="inbound_12345678"><br> <condition field="destination_number" expression="^(12345678)$"><br> <action application="bridge" data="user/myuserid"/><br></div><div> </condition><br></div><div></extension><br><br></div><div>Thanks<br></div><div><br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-02-02 19:16 GMT+01:00 Sergey Safarov <span dir="ltr"><<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">What is way you planing to use for link DID with user?<div><br></div><div>Sergey</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Feb 2, 2016 at 8:20 PM, Marc S <span dir="ltr"><<a href="mailto:therebel22@gmail.com" target="_blank">therebel22@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div><div><div><div><div>Hello,<br></div>sorry for asking again :<br><br></div>I hav an asterisk that register on Freeswitch (as a user). <br><br></div>When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension.<br><br></div>Here my bridge tests<span><br><br><action application="bridge" data="user/myuserid"/> <br><br></span></div>=> s extension in asterisk instead of extension<br><div><br><action application="bridge" data="sofia/mycontext/myextension@<ASTERISK_IP>"/><br><br></div><div>=> Not authenticated in asterisk (because no IP authentication in asterisk)<br><br></div><div>Have you an idea how to send real extension instead of s extension ?<br></div><div>Thanks<br></div><div><br><br></div><div><br><br><div><br><br></div></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2016-01-03 10:45 GMT+01:00 Marc S <span dir="ltr"><<a href="mailto:therebel22@gmail.com" target="_blank">therebel22@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div><div>Hello,<br><br></div>i'm discovering FS. I hav read a lot about users and gateways.<br><br></div>I would like to FS act as registrar for authenticated SIP trunking.<br><br></div>- Customers IPBX would register with login/password to Freeswitch. <br>- Incoming call would be routed to these SIP trunks in dialplan XML.<br><br></div>directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id :<br><br><action application="bridge" data="user/myuserid"/><br><br></div>gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar.<br><br></div>Is it possible to FS to act as authenticated SIP trunking registrar ?<br><br></div>Thanks a lot,<br></div>Marc<br><div><div><div><div><br><br></div></div></div></div></div>
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