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

[Mirko Brankovic]

>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 at 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 at 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
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