Asterisk and GrandstreamHT503 caller ID Issue

Dear All,

Am facing a Problem to get caller id in grandstream to Asterisk;
Ealier i got it comes in P-asserted identity…
now its not comming and not getting caller id
please help me

Configuration

GrandstreamHT503

FXO Port
Primary server IP:xxx.xxx.x.x
usr ID:153
secret:xxx
port:5062
Caller ID Scheme:ETSI-DTMF during ringing(INDIA)
FSK Caller ID Minimum RX Level (dB): 0
Caller ID Transport Type:Relay via SIP Assereted Identity
Impedance-based:COMPLEX3 370+620ohms||310nf)
Number of Rings:4
PSTN Ring Thru FXS;NO
PSTN Ring Thru Delay (sec):2
Stage Method (1/2):1

==================
sip.conf

[153]
deny=0.0.0.0/0.0.0.0
secret=153abc
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/153
mailbox=153@device
permit=0.0.0.0/0.0.0.0
callerid=153 <153>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

=========
extension.conf

[from-ht503]
exten => _XX.,1,NoOp$(${CALLERID(number)}
exten => _XX.,2,Goto(from-pstn,1)

While making a call,call received in asterisk but not getting Caller ID

Logs

<— SIP read from UDP:192.168.1.193:38848 —>
INVITE sip:123@192.168.1.198:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK1256157716;rport
Route: sip:192.168.1.198:5060;lr
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 20 INVITE
Contact: sip:153@192.168.1.193:38848
Max-Forwards: 70
User-Agent: Grandstream HT-503 V2.0A 1.0.12.5 chip V2.2
Privacy: none
P-Asserted-Identity: sip:153@192.168.1.198
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 454

v=0
o=153 8002 8000 IN IP4 192.168.1.193
s=SIP Call
c=IN IP4 192.168.1.193
t=0 0
m=audio 23784 RTP/AVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
— (17 headers 20 lines) —
Sending to 192.168.1.193:38848 (no NAT)
Sending to 192.168.1.193:38848 (no NAT)
Using INVITE request as basis request - 1063945026-38848-3@BJC.BGI.B.BJD
Found peer ‘153’ for ‘153’ from 192.168.1.193:38848

<— Reliably Transmitting (no NAT) to 192.168.1.193:38848 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK1256157716;received=192.168.1.193;rport=38848
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060;tag=as2de32fab
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 20 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="691566dd"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '1063945026-38848-3@BJC.BGI.B.BJD’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.193:38848 —>
ACK sip:123@192.168.1.198:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK1256157716;rport
Route: sip:192.168.1.198:5060;lr
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060;tag=as2de32fab
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 20 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.193:38848 —>
INVITE sip:123@192.168.1.198:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK984053011;rport
Route: sip:192.168.1.198:5060;lr
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 21 INVITE
Contact: sip:153@192.168.1.193:38848
Authorization: Digest username=“153”, realm=“asterisk”, nonce=“691566dd”, uri=“sip:123@192.168.1.198:5060”, response=“d21008b07a0bd10ba6072b1a4e408782”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V2.0A 1.0.12.5 chip V2.2
Privacy: none
P-Asserted-Identity: sip:153@192.168.1.198
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 454

v=0
o=153 8002 8000 IN IP4 192.168.1.193
s=SIP Call
c=IN IP4 192.168.1.193
t=0 0
m=audio 23784 RTP/AVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
— (18 headers 20 lines) —
Sending to 192.168.1.193:38848 (no NAT)
Using INVITE request as basis request - 1063945026-38848-3@BJC.BGI.B.BJD
Found peer ‘153’ for ‘153’ from 192.168.1.193:38848
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found unknown media description format G729E for ID 102
Found unknown media description format AAL2-G726-16 for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.193:23784
Looking for 123 in from-internal (domain 192.168.1.198)
list_route: hop: sip:153@192.168.1.193:38848

<— Transmitting (no NAT) to 192.168.1.193:38848 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK984053011;received=192.168.1.193;rport=38848
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 21 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:123@192.168.1.198:5060
Content-Length: 0

<------------>
– Executing [123@from-internal:1] NoOp(“SIP/153-00000003”, “”" <153>") in new stack
– Executing [123@from-internal:2] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:3] NoOp(“SIP/153-00000003”, “1”) in new stack
– Executing [123@from-internal:4] NoOp(“SIP/153-00000003”, “iso8859-1”) in new stack
– Executing [123@from-internal:5] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:6] NoOp(“SIP/153-00000003”, “153”) in new stack
– Executing [123@from-internal:7] NoOp(“SIP/153-00000003”, “1”) in new stack
– Executing [123@from-internal:8] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:9] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:10] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:11] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:12] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:13] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:14] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:15] NoOp(“SIP/153-00000003”, “”" <>") in new stack
– Executing [123@from-internal:16] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:17] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:18] NoOp(“SIP/153-00000003”, “iso8859-1”) in new stack
– Executing [123@from-internal:19] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:20] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:21] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:22] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:23] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:24] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:25] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:26] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:27] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:28] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:29] NoOp(“SIP/153-00000003”, “”" <153>") in new stack
– Executing [123@from-internal:30] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:31] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:32] NoOp(“SIP/153-00000003”, “iso8859-1”) in new stack
– Executing [123@from-internal:33] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:34] NoOp(“SIP/153-00000003”, “153”) in new stack
– Executing [123@from-internal:35] NoOp(“SIP/153-00000003”, “1”) in new stack
– Executing [123@from-internal:36] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:37] NoOp(“SIP/153-00000003”, “allowed_not_screened”) in new stack
– Executing [123@from-internal:38] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:39] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:40] NoOp(“SIP/153-00000003”, “123”) in new stack
– Executing [123@from-internal:41] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:42] NoOp(“SIP/153-00000003”, “”) in new stack
– Executing [123@from-internal:43] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:44] NoOp(“SIP/153-00000003”, “0”) in new stack
– Executing [123@from-internal:45] NoOp(“SIP/153-00000003”, “0”) in new stack
– Auto fallthrough, channel ‘SIP/153-00000003’ status is ‘UNKNOWN’
Scheduling destruction of SIP dialog '1063945026-38848-3@BJC.BGI.B.BJD’ in 6400 ms (Method: INVITE)

<— Reliably Transmitting (no NAT) to 192.168.1.193:38848 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK984053011;received=192.168.1.193;rport=38848
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060;tag=as03b46cc0
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 21 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.193:38848 —>
ACK sip:123@192.168.1.198:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.193:38848;branch=z9hG4bK984053011;rport
Route: sip:192.168.1.198:5060;lr
From: sip:153@192.168.1.198;tag=766277840
To: sip:123@192.168.1.198:5060;tag=as03b46cc0
Call-ID: 1063945026-38848-3@BJC.BGI.B.BJD
CSeq: 21 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog '1063945026-38848-3@BJC.BGI.B.BJD’ Method: ACK