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 <http://www.inextrix.com>
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