Freeswitch as Registrar for SIP Trunk
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
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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Post you register and incoming sip peer settings from yor asterisk if you want to got further help. 2016-02-02 22:06 GMT+02:00 Marc S <therebel22@gmail.com>:
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 > > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http:// lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Look at REGISTER messages from asterisk. They have header Contact: s@<asterisk_ip> You get what you ask for :-) On Tuesday 02 February 2016 21:06:01 Marc S 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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________________________________ __ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 > > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http:// lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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"..
On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks for all replies. FS would be used as SBC. I would like my customers' IPBX (Asterisk, Alcatel, ..) register against FS, as a SIP trunk and then i would like routing several DID (not only s) on incoming call to FS to customers IPBX with bridge action. I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, because of security. But registring user in FS seems to expect SIP phone only as user to register, not IPBX as user : Here my asterisk Register : REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport From: <sip:testipbx2@1.2.3.4>;tag=as47a2d14b To: <sip:testipbx2@1.2.3.4> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf@127.0.1.1 CSeq: 103 REGISTER User-Agent: Asterisk Max-Forwards: 70 Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", nc=00000001 Expires: 120 Contact: <sip:s@5.6.7.8> Event: registration Content-Length: 0 Here FS->Asterisk invite on incoming call INVITE sip:s@5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c Max-Forwards: 69 From: "45678776" <sip:45678776@1.2.3.4>;tag=4v1S65HQ930mF To: <sip:s@5.6.7.8> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 CSeq: 86886190 INVITE Contact: <sip:mod_sofia@1.2.3.4:5060> User-Agent: FS/1.6 .. Thanks 2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:
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"..
On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You seem like you're trying to fly before you can crawl with FreeSWITCH. You have some base level technological understanding to acquire prior to getting to the end results. You probably didn't read what I wrote in my previous email. Your device is registering as sip:s@5.6.7.8, so we're gonna follow the rules and call you back at sip:s@5.6.7.8 You can however in FreeSWITCH force the auth user to match the user and there by having exactly one DID per sip account, or you could change the RURI (which an example of how to do this is in the vanilla config we ship) Here it is: <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri _user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/> <action application="execute_extension" data="ruri"/> </condition> </extension> On Tue, Feb 2, 2016 at 2:42 PM, Marc S <therebel22@gmail.com> wrote:
Thanks for all replies.
FS would be used as SBC.
I would like my customers' IPBX (Asterisk, Alcatel, ..) register against FS, as a SIP trunk and then i would like routing several DID (not only s) on incoming call to FS to customers IPBX with bridge action.
I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, because of security.
But registring user in FS seems to expect SIP phone only as user to register, not IPBX as user :
Here my asterisk Register :
REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport From: <sip:testipbx2@1.2.3.4>;tag=as47a2d14b To: <sip:testipbx2@1.2.3.4> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf@127.0.1.1 CSeq: 103 REGISTER User-Agent: Asterisk Max-Forwards: 70 Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", nc=00000001 Expires: 120 Contact: <sip:s@5.6.7.8> Event: registration Content-Length: 0
Here FS->Asterisk invite on incoming call
INVITE sip:s@5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c Max-Forwards: 69 From: "45678776" <sip:45678776@1.2.3.4>;tag=4v1S65HQ930mF To: <sip:s@5.6.7.8> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 CSeq: 86886190 INVITE Contact: <sip:mod_sofia@1.2.3.4:5060> User-Agent: FS/1.6 .. Thanks
2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:
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"..
On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Thanks a lot for reply ! Sorry for misunderstanding. I'm going to study what you wrote. 2016-02-02 21:50 GMT+01:00 Brian West <brian@freeswitch.org>:
You seem like you're trying to fly before you can crawl with FreeSWITCH. You have some base level technological understanding to acquire prior to getting to the end results. You probably didn't read what I wrote in my previous email. Your device is registering as sip:s@5.6.7.8, so we're gonna follow the rules and call you back at sip:s@5.6.7.8
You can however in FreeSWITCH force the auth user to match the user and there by having exactly one DID per sip account, or you could change the RURI (which an example of how to do this is in the vanilla config we ship)
Here it is:
<!--
This is an example of how to override the RURI on an outgoing invite to a registered contact.
-->
<extension name="ruri">
<condition field="destination_number" expression="^ruri$">
<action application="bridge" data="sofia/${ruri_profile}/${ruri _user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
</condition>
</extension>
<extension name="7004">
<condition field="destination_number" expression="^7004$">
<action application="set" data="ruri_profile=default"/>
<action application="set" data="ruri_user=2000"/>
<action application="set" data="ruri_contact=1001@${domain_name}"/>
<action application="execute_extension" data="ruri"/>
</condition>
</extension>
On Tue, Feb 2, 2016 at 2:42 PM, Marc S <therebel22@gmail.com> wrote:
Thanks for all replies.
FS would be used as SBC.
I would like my customers' IPBX (Asterisk, Alcatel, ..) register against FS, as a SIP trunk and then i would like routing several DID (not only s) on incoming call to FS to customers IPBX with bridge action.
I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, because of security.
But registring user in FS seems to expect SIP phone only as user to register, not IPBX as user :
Here my asterisk Register :
REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport From: <sip:testipbx2@1.2.3.4>;tag=as47a2d14b To: <sip:testipbx2@1.2.3.4> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf@127.0.1.1 CSeq: 103 REGISTER User-Agent: Asterisk Max-Forwards: 70 Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", nc=00000001 Expires: 120 Contact: <sip:s@5.6.7.8> Event: registration Content-Length: 0
Here FS->Asterisk invite on incoming call
INVITE sip:s@5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c Max-Forwards: 69 From: "45678776" <sip:45678776@1.2.3.4>;tag=4v1S65HQ930mF To: <sip:s@5.6.7.8> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 CSeq: 86886190 INVITE Contact: <sip:mod_sofia@1.2.3.4:5060> User-Agent: FS/1.6 .. Thanks
2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:
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"..
On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:
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 > > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http:// lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
We used LUA script to deal with registered asterisk systems. Example DID 888 555 6666 is assigned to user 101, registered on some ip say 8.8.8.8 Contact will look like s@8.8.8.8 or 101@8.8.8.8.8 LUA Code DIALED_NUMBER=8885556666 USER = 111 location = api:execute("sofia_contact", USER) or "" destinationlocation=location:gsub("sip:(.-)@","sip:"..DIALED_NUMBER.."@") -- This will replace user part in contact heard, (s or even username) with the DID number. -- destinationlocation will look like sip:8885556666@8.8.8.8 if (destinationlocation ~= "") then session:execute("bridge","{sip_cid_type=rpid,loop=3}"..destinationlocation) end Hope this help. *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 On Tue, Feb 2, 2016 at 12:29 PM, Oz Mortimer <omortimer@gmail.com> wrote:
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"..
On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Read theread "Conectin a SPA3102 as pstn gateway on FS <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115045.html> " {sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/ reg_user@mydomain.org If requred mutal autertication {sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/ reg_user@mydomain.org On Tue, Feb 2, 2016 at 10:03 PM, Marc S <therebel22@gmail.com> wrote:
I want to use mod xml curl to generate dynamic dialplan xml like this :
<extension name="inbound_12345678"> <condition field="destination_number" expression="^(12345678)$"> <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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I will try soon, thanks 2016-02-03 7:42 GMT+01:00 Sergey Safarov <s.safarov@gmail.com>:
Read theread "Conectin a SPA3102 as pstn gateway on FS <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115045.html> "
{sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/ reg_user@mydomain.org
If requred mutal autertication
{sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/ reg_user@mydomain.org
On Tue, Feb 2, 2016 at 10:03 PM, 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (7)
-
Alexandr Usov
-
Brian West
-
Jai Rangi
-
Marc S
-
Oz Mortimer
-
Sergey Okhapkin
-
Sergey Safarov