From what I see BYE is comming from your Asterisk Vicidial box:

1. U 60.80.70.250:5060 -> 92.42.132.242:5060 2. BYE sip:18177779695@92.42.132.242:5060;transport=udp SIP/2.0. So I guess you need to see what is happening there, since call set up was correct. I see the From is not the same. Plus, I don't think it is safe to place all those details like that online, like Authenticated Invite,.... Regards, mirko On Fri, Sep 16, 2016 at 2:23 PM, devang nathwani < devang.nathwani31589@gmail.com> wrote:

Hello,

I am trying to call from vicidial to freeswitch(billing platform) to provider(gateway) so call flow would be like this

My vicidial 60.80.70.250

my freeswitch 92.42.132.242

my provider(gateway) 72.2.242.42

I am dialing 18177779695

When i try using dialtype 'MANUAL' the call is routing correctly without any issue. But when i try using dialtype 'RATIO', i am facing issue when freeswitch is resending the provider(gateway)'s response to vicidial the asterisk of vicidial server responding with X-Asterisk-HangupCause: Unknown. X-Asterisk-HangupCauseCode: 0.

I am attaching the sip log of both working(MANUAL) and and not working(RATIO) scenario here. Please note that the attached sip log is from freeswitch server(billing platform).

Working sip log is here http://pastebin.com/ma4e3xzk

Not working sip log is here http://pastebin.com/6PAnyDBc

Please advice, what am I missing here?

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-- Regards, Mirko