i integrate asterisk with softswitch provider via NSN softswitch. Every no answer PSTN call (that using SS7) route by NSN to asterisk server. But NSN not completed diversion field, so RDNIS variable filled as Anonymous. Anyone have same experience with this problem …? 
my sip.conf
[trunk-softswitch]
type=friend
host=10.9.9.3
qualify=yes
dtmfmode=auto
canreinvite=no
context=context-softswitch
sendrpid=yes
example sip debug:
<— SIP read from UDP:10.9.9.3:5060 —>
INVITE sip:20002@10.9.9.8.6:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.9.9.3:5060;branch=z9hG4bK20502574190445994947.
From: sip:4514735889@anonymous.invalid;user=phone;tag=507B7514-4EAFBE45-AC1B28AA
To: sip:20002@anonymous.invalid:5060;user=phone
Call-ID: 507B7514-072FC325@hiepme243
CSeq: 1 INVITE
Accept: application/sdp,application/isup,multipart/mixed,application/vnd.siemens.key-event,application/vnd.siemens.surpass,application/dtmf-relay
Contact: sip:10.9.9.3:5060
Diversion: sip:Anonymous@Anonymous.invalid;reason="no-answer"
MIME-Version: 1.0
Supported: timer,100rel
Max-Forwards: 70
P-Asserted-Identity: sip:4514735889@anonymous.invalid;user=phone
Session-Expires: 1800
Allow: ACK,INFO,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,PRACK,UPDATE
Content-Type: application/sdp
Content-Length: 380
…