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

[Nick Khamis]

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

UA- 192.168.2.11
OpenSIPS: 192.168.2.5
Asterisk (Test Box): 192.168.2.10

A slightly unrelated, do we have to "force_rport();" on all SIP
traffic or only the initial
INVITE?

Thanks in Advance, and great product!!!

Nick.