[freeswitch]type=peerhost=IP_ADDRESS_OF_FREESWITCH_SERVERusername=HOSTNAME.DOMAIN.COMport=5080fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVERsecret=BOOTH_WAY_PASSWORDHi MichaelThank 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 PatelOn 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:-MSCOn 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-extWe want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?--
Thanks,
Rutu PatelOn 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/AsteriskWith kind regards,JurijsOn 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.
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