<div dir="ltr">This is because Asterisk will register as s@ on the contact when it registers to FreeSWITCH, so you can fix this by fixing your register line in sip.conf on Asterisk to register as 12345678 vs the default s<div><br></div><div>If I recall its this format:</div><div><br></div><div>
<p class=""><span class="">register => user[:secret[:authuser]]@host[:port][/extension]</span></p><p class=""><span class=""><br></span></p><p class=""><span class="">Thanks,</span></p></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 2, 2016 at 2:06 PM, Marc S <span dir="ltr"><<a href="mailto:therebel22@gmail.com" target="_blank">therebel22@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 dir="ltr"><div><div><div><div><div><div>Thanks, Asterisk is already registred to FS.<br><br></div>But when incoming call to FS is bridged to registered Asterisk :<span class=""><br><br><span> <action application="bridge" data="user/myuserid"/><br><br></span></span></div><span>FS send SIP Message to asterisk :<br><br></span></div><span>INVITE s@<asterisk_ip><br><br></span></div><span>instead of :<br><br></span></div><span>INVITE </span>12345678@<span><asterisk_ip><br><br></span></div><span>I would like to get </span>12345678 in asterisk..<br><br><div><div><div><div><div><div><span><br><br></span></div></div></div></div></div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2016-02-02 20:57 GMT+01:00 Alexandr Usov <span dir="ltr"><<a href="mailto:blessendor@gmail.com" target="_blank">blessendor@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>You must have in Asterisk configs something as in my examples. <br><br>;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. With static IP of both FS and Asterisk you not need to do register from asterisk.<br>register => asterisk2fs:strongpassword@ip_of_freeswitch/freeswitch<br><br><br>;; type=friend means that this peer can be used for outbound and inbound calls, so we don't need to create two peer settings block ( [freeswitch-in] and [freeswitch-out] )<br><br>[freeswitch] ;; you maybe want to use here your public DID number, as well as in username/fromuser settings<br>type=friend<br>username=asterisk2fs<br>fromuser=asterisk2fs<br>fromdomain=ip_of_freeswitch<br>host=ip_of_freeswitch<br>context=from-freeswitch<br>secret=strongpassword<br>insecure=port,invite<br>qualify=yes<br>port=5060<br><br><br></div>Your context [from-freeswitch] must have an extension, named as 'freeswitch' (or your DID number) for incoming calls operations.<br><br><br><br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2016-02-02 21:28 GMT+02:00 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">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><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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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font size="1" face="courier new, monospace"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br></font></p><p><font size="2" face="monospace, monospace"><b><i>Twitter: @FreeSWITCH , @briankwest</i></b><br><a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br><a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p><p><font face="monospace, monospace">Got Bugs? Report them <a href="https://freeswitch.org/jira" target="_blank">here</a>! | Reddit: <a href="https://www.reddit.com/r/freeswitch" target="_blank">/r/freeswitch</a></font></p>
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