I think you need to explain what you are trying to achieve..What you are doing currently is registering a user against freeswitch - not dissimilar to logging in to Skype.In your bridge statement you are calling the registered user - again like you would in Skype.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"..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 ?2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway.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".Google is your friend..I want to use mod xml curl to generate dynamic dialplan xml like this :
<extension name="inbound_12345678">
<condition field="destination_number" expression="^(12345678)$">
<action application="bridge" data="user/myuserid"/></condition></extension>Thanks2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov@gmail.com>:What is way you planing to use for link DID with user?SergeyOn Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22@gmail.com> wrote:_________________________________________________________________________=> s extension in asterisk instead of extensionHere my bridge testsWhen a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension.I hav an asterisk that register on Freeswitch (as a user).Hello,sorry for asking again :
<action application="bridge" data="user/myuserid"/>
<action application="bridge" data="sofia/mycontext/myextension@<ASTERISK_IP>"/>=> Not authenticated in asterisk (because no IP authentication in asterisk)Have you an idea how to send real extension instead of s extension ?Thanks2016-01-03 10:45 GMT+01:00 Marc S <therebel22@gmail.com>:MarcThanks a lot,Is it possible to FS to act as authenticated SIP trunking registrar ?gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar.directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id :- Customers IPBX would register with login/password to Freeswitch.I would like to FS act as registrar for authenticated SIP trunking.Hello,i'm discovering FS. I hav read a lot about users and gateways.
- Incoming call would be routed to these SIP trunks in dialplan XML.
<action application="bridge" data="user/myuserid"/>
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