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 If I recall its this format: register => user[:secret[:authuser]]@host[:port][/extension] Thanks, On Tue, Feb 2, 2016 at 2:06 PM, Marc S <therebel22@gmail.com> wrote:
Thanks, Asterisk is already registred to FS.
But when incoming call to FS is bridged to registered Asterisk :
<action application="bridge" data="user/myuserid"/>
FS send SIP Message to asterisk :
INVITE s@<asterisk_ip>
instead of :
INVITE 12345678@<asterisk_ip>
I would like to get 12345678 in asterisk..
2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor@gmail.com>:
You must have in Asterisk configs something as in my examples.
;;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=5060
Your context [from-freeswitch] must have an extension, named as 'freeswitch' (or your DID number) for incoming calls operations.
2016-02-02 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..
On 2 Feb 2016, at 19:03, Marc S <therebel22@gmail.com> wrote:
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>
Thanks
2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov@gmail.com>:
What is way you planing to use for link DID with user?
Sergey
On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22@gmail.com> wrote:
Hello, sorry for asking again :
I hav an asterisk that register on Freeswitch (as a user).
When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension.
Here my bridge tests
<action application="bridge" data="user/myuserid"/>
=> s extension in asterisk instead of extension
<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 ? Thanks
2016-01-03 10:45 GMT+01:00 Marc S <therebel22@gmail.com>:
> Hello, > > i'm discovering FS. I hav read a lot about users and gateways. > > I would like to FS act as registrar for authenticated SIP trunking. > > - Customers IPBX would register with login/password to Freeswitch. > - Incoming call would be routed to these SIP trunks in dialplan XML. > > directory/users does not seem to be the solution because in > dialplan, destination DID can't be defined, only user id : > > <action application="bridge" data="user/myuserid"/> > > gateway seems to be designed for SIP trunking to remote SIP gateway, > not for FS to act as registrar. > > Is it possible to FS to act as authenticated SIP trunking registrar ? > > Thanks a lot, > Marc > > >
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