[OpenSIPS-Users] 100 Giving a try Not getting routed to Asterisk

dotnetdub dotnetdub at gmail.com
Sun Jan 6 22:18:19 CET 2013

On 6 January 2013 20:39, Nick Khamis <symack at gmail.com> wrote:
> Hello Everyone,
> After getting out Asterisk machines up and running, two way audio
> etc... We would like to put an OpenSIPS server between the world and
> our Asterisk boxes as one would imagine. The initial INVITE is getting
> routed correctly however, the "Giving a try" from our SIP trunk is not
> making it's way to the Asterisk box.
> I have attached some pastebins of:
> Asterisk SIP Log: http://pastebin.com/VdtAKBH9
> OpenSIPS Log: http://pastebin.com/BLcgFrV5
> OpenSIPS Debug (Console): http://pastebin.com/gN6hgsxz

I don't think we are seeing the full picture here... What is ?

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