Hi! I want that my FeeSWITCH receives calls from my opensips' users without requiring authentication. In other words, I want that my FS receives unauthenticated calls, only from the IP address of my OpenSIPS server. any help, please???
Hi. You can use acl for that. <http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.html> 2012/4/16 azza miled <azza.miled@gmail.com>:
Hi! I want that my FeeSWITCH receives calls from my opensips' users without requiring authentication. In other words, I want that my FS receives unauthenticated calls, only from the IP address of my OpenSIPS server. any help, please???
-- Mit freundlichen Grüßen *Karsten Horsmann*
Thanks for the response, but evec with these steps, I still have the error "call failed:proxy authentication required" when I dial the IVR extension from an opensips' phone. Do any one know, what shall I do??? 2012/4/16 Karsten Horsmann <khorsmann@gmail.com>
Hi.
You can use acl for that.
< http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.htm...
2012/4/16 azza miled <azza.miled@gmail.com>:
Hi! I want that my FeeSWITCH receives calls from my opensips' users without requiring authentication. In other words, I want that my FS receives unauthenticated calls, only from the IP address of my OpenSIPS server. any help, please???
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
On Mon, Apr 16, 2012 at 8:36 AM, azza miled <azza.miled@gmail.com> wrote:
Thanks for the response, but evec with these steps, I still have the error "call failed:proxy authentication required" when I dial the IVR extension from an opensips' phone. Do any one know, what shall I do???
Do you need SIP authentication between your FS box and your OpenSIPS box? I'm guessing not. If that's the case then I recommend opening up this file:
/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml Find the tag with this opening line: <list name="domains" default="deny"> Add a new line in there like this: <node type="allow" cidr="x.x.x.x/32"/> where x.x.x.x is the IP address of your OpenSIPS server. then go to fs_cli and issue this command: reloadacl Lastly, you'll need to route the inbound calls from OpenSIPS through your 'public' dialplan context because that's where ACL-allowed inbound calls normally get processed. You can either create your IVRs in the public context or you can transfer the inbound call over to the default context (or whatever context you are using.) -MC
Thanks a lot Michael.I am a real begginer in the SIP world and your response was very helpful. It worked :))) Thanks again. 2012/4/16 Michael Collins <msc@freeswitch.org>
On Mon, Apr 16, 2012 at 8:36 AM, azza miled <azza.miled@gmail.com> wrote:
Thanks for the response, but evec with these steps, I still have the error "call failed:proxy authentication required" when I dial the IVR extension from an opensips' phone. Do any one know, what shall I do???
Do you need SIP authentication between your FS box and your OpenSIPS box? I'm guessing not. If that's the case then I recommend opening up this file:
/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml
Find the tag with this opening line: <list name="domains" default="deny">
Add a new line in there like this: <node type="allow" cidr="x.x.x.x/32"/> where x.x.x.x is the IP address of your OpenSIPS server.
then go to fs_cli and issue this command: reloadacl
Lastly, you'll need to route the inbound calls from OpenSIPS through your 'public' dialplan context because that's where ACL-allowed inbound calls normally get processed. You can either create your IVRs in the public context or you can transfer the inbound call over to the default context (or whatever context you are using.)
-MC
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
On Tue, Apr 17, 2012 at 1:17 AM, azza miled <azza.miled@gmail.com> wrote:
Thanks a lot Michael.I am a real begginer in the SIP world and your response was very helpful. It worked :))) Thanks again.
Thanks for checking in. Don't forget to buy the books that I co-authored with several other really smart guys. :) The bridge<http://www.amazon.com/FreeSWITCH-1-0-6-Anthony-Minessale/dp/1847199968/ref=sr_1_1?ie=UTF8&qid=1334678891&sr=8-1>book and the cookbook<http://www.amazon.com/FreeSWITCH-Cookbook-Anthony-Minessale/dp/1849515409/ref=sr_1_2?ie=UTF8&qid=1334678891&sr=8-2> . -MC
participants (3)
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azza miled
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Karsten Horsmann
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Michael Collins