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

Nick Khamis symack at gmail.com
Mon Jan 7 00:22:31 CET 2013

Thank you so much for your response. I should have mentioned that I
changed IPs, phone numbers, and domain names to bogus values. Never
know who's reading, and I would not do that to anyone either.

Irregardless, it turned out that setting nat=yes in  sip.conf solved
the problem.



On 1/6/13, dotnetdub <dotnetdub at gmail.com> wrote:
> 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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