<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<br><br><action application="bridge" data="user/myuserid"/> <br><br></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 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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