Weird problem with sip and nat

[grab3]

Hi! I've got some weird problem. I have Asterisk box behind pfsense. Forwarded 5060 and rtp(10000-20000) ports to my

asterisk box. Asterisk registering with sip provider fine, can call through prov with no problems. But when

providers support or somebody else from internal sipnet.ru's number is trying to call me, call fails. As you can see

in attachement, it fails because INVITE packets by some weird reason are not going through pf sense. It is even more

strange because i have remote office connected through my asterisk and they can call me with no problems, their

packets are going through pfsense fine.
I have no rules which could block provs address. And it is obvious, cus i can call through them.

What could be the problem?

Anton

PS outer interface is MELS(em1)
txt files are actually a cap files, it can beopened in wireshark
it is samples of good conversation between remote office and asterisk,
and bad conversation between asterisk and provider.










captureinneriface_em0.txt
captureouteriface_em1.txt
captureinneriface_good_em0.txt
captureouteriface_good_em1.txt