register => user[:secret[:authuser]]@host[:port][/extension]
Thanks,
I would like to get 12345678 in asterisk..INVITE 12345678@<asterisk_ip>instead of :INVITE s@<asterisk_ip>FS send SIP Message to asterisk :Thanks, Asterisk is already registred to FS.But when incoming call to FS is bridged to registered Asterisk :
<action application="bridge" data="user/myuserid"/>2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor@gmail.com>:You must have in Asterisk configs something as in my examples.Your context [from-freeswitch] must have an extension, named as 'freeswitch' (or your DID number) for incoming calls operations.
;;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.
register => asterisk2fs:strongpassword@ip_of_freeswitch/freeswitch
;; 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] )
[freeswitch] ;; you maybe want to use here your public DID number, as well as in username/fromuser settings
type=friend
username=asterisk2fs
fromuser=asterisk2fs
fromdomain=ip_of_freeswitch
host=ip_of_freeswitch
context=from-freeswitch
secret=strongpassword
insecure=port,invite
qualify=yes
port=50602016-02-02 21:28 GMT+02:00 Marc S <therebel22@gmail.com>: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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Official FreeSWITCH Sites
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http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
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Official FreeSWITCH Sites
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_________________________________________________________________________
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Brian West
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