[Freeswitch-users] call is not routing from asterisk->freeswitch(billing platform)->provider

[devang nathwani]

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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