Extesion check after a SIP REFER?

Hi all,

As I could understand, looks like after each REFER, Asterisk checks
whether the transferor can reach the target extension via its default
context. See bellow an example:

REFER sip:235 <at> 200.XXX.XXX.35 SIP/2.0
Via: SIP/2.0/UDP 200.YYY.YYY.87;branch=z9hG4bK76bec12d2FD069B6
From: <sip:10.ext101 <at> 200.YYY.YYY.87>;tag=F810E8F1-6A9966A
To: "EXT 1" <sip:235 <at> 200.XXX.XXX.35>;tag=as638c860f
CSeq: 2 REFER
Call-ID: 376487393bb717380578a58b3ad9707c <at> 200.XXX.XXX.35
Contact: <sip:10.ext101 <at> 200.YYY.YYY.87>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.2.0972
Refer-To:
<sip:2 <at> servername.domain.com.br;user=phone?Replaces=fac455c9-735967e2-bf
510297%40200.YYY.YYY.87%3Bto-tag%3Das39a366f4%3Bfrom-tag%3D840EBF78-DF8C
E1D5>
Referred-By: <sip:10.ext101 <at> servername.domain.com.br>
Max-Forwards: 70
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Call 376487393bb717380578a58b3ad9707c <at> 200.XXX.XXX.35
376487393bb717380578a58b3ad9707c <at> 200.XXX.XXX.35  got a SIP call transfer
from caller: (REFER)!
SIP transfer to extension 2 <at> 10.profile_controller
2 <at> 10.profile_controller by 10.ext101 <at> servername.domain.com.br