Hi,

Thank you for your help to debug the issue.

We are now trying without opensips but still facing authentication issue.

We created two users of freeswitch in asterisk as peer.

For example, First peer in sip.conf is 456789 and second is 123456.

[123456]
type=friend
host=x.x.x.45
username=123456
secret=abcdef
disallow=all
allow=g729
trustrpid=yes
canreinvite=yes
sendrpid=yes
context=extensions-test
qualify=yes
directmedia=yes

Now when we are trying to reach 123456, the request reach to asterisk but not authenticate proper user(123456). It authenticate first peer(456789) defined in sip.conf.

If we define 123456 before 456789 then it works but we have multiple peers in asterisk so how can we manage that ?

We also tried by sending user credential direct in bridge,but still it is authenticate the first peer defined in sip.conf .

{sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER@IP
 
​Can anyone please explain how to authenticate specific peer ?

Please let me know if you want any other details.​

--
Thanks,
Rutu Patel
​​


On Fri, Apr 1, 2016 at 3:56 PM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi,

I'm not sure about your environment, but on this point I will recommend you to isolate issue. So on this point you should understand if your sip trunk between Asterisk and Freeswitch configured correctly, without involving Opensips and etc. To test SIP trunk you can try to send call directly from freeswitch to Asterisk and see if it works, you can send call just directly from freeswitch console, something similar to:
originate sofia/example/300@foo.com 8600
https://freeswitch.org/confluence/display/FREESWITCH/mod_commands

If you call will reach Asterisk without any issue, then this means that sip trunk created correctly and issue is somewhere on Opensips side or any other part of your set-up.

Second option will be to try to configure sip trunk without password and check if it helps

https://freeswitch.org/confluence/display/FREESWITCH/Asterisk

With kind regards,

Jurijs

On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Juris, 

Yes we added peer in asterisk sip.conf.

Are we still missing any configurations? 


On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,

Just a shot in a dark. :)

Did you added sip peer record in sip.conf, something similar to:

[freeswitch]
type=peer
host=IP_ADDRESS_OF_FREESWITCH_SERVER
port=5080
fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
secret=BOOTH_WAY_PASSWORD

With kind regards,


Jurijs

On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Michael

Thank you for your assistance.

We are trying to make call between two freeswitch users.

One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.

In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.

I am attaching some more logs if anyone can guide in right direction.

--
Thanks,
Rutu Patel



On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:

-MSC

On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,

Thank you for reply.

We have registered extension of Freeswitch in Asterisk as below

register=>test-ext:secret@192.168.1.64/test-ext

​We want to send call from Freeswitch to Asterisk​.

What authentication parameters we need to set in Freeswitch ?

--
Thanks,
Rutu Patel
On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls.
The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
 



On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,

In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...

First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...

https://freeswitch.org/confluence/display/FREESWITCH/Asterisk

With kind regards,

Jurijs

On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,

We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
 
Call flow is as below:

User => opensips => FS => PBX,

but call not sending out from FS and drop by FS.

Here I am attaching both sip logs and freeswitch logs.

Can anyone please help to sort out the issue.
--
Thanks,
Rutu Patel

_________________________________________________________________________
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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
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_________________________________________________________________________
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consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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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
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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
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--
--
Thanks,
Rutu Patel
iNextrix Technologies Pvt. Ltd.
www.inextrix.com


_________________________________________________________________________
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
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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
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