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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_________________________________________________________________________
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