SRTP on Polycom phone not working please help

Installed Asterisk asterisk-1.8.26.1 with freePBX 2.11,

Configured SRTP and it works good on softphone(blink)but the polycom phone doesn’t make outgoing call when SRTP enabled under extension “encryption Yes(SRTP only)”

here are the astrisk debug logs.

PBX1CLI>
PBX1
CLI>
PBX1CLI>
PBX1
CLI>
PBX1CLI>
PBX1
CLI>
PBX1*CLI>
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
Reliably Transmitting (NAT) to 192.168.1.143:56951:
OPTIONS sip:[email protected]:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.28:5061;branch=z9hG4bK0bb01d3e;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4a57b32b
To: sip:[email protected]:5061;transport=tls
Contact: sip:[email protected]:5061;transport=TLS
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.26.1)
Date: Tue, 15 Apr 2014 19:00:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from TLS:192.168.1.143:56951 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.28:5061;rport=5061;received=192.168.1.28;branch=z9hG4bK0bb01d3e
Call-ID: [email protected]:5061
From: “Unknown” sip:[email protected];tag=as4a57b32b
To: sip:[email protected];tag=z9hG4bK0bb01d3e
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Accept: application/sdp, application/conference-info+xml, application/simple-message-summary, multipart/related, application/rlmi+xml, application/dialog-info+xml, multipart/related, application/rlmi+xml, application/pidf+xml, application/watcherinfo+xml, application/xcap-diff+xml, application/watcherinfo+xml, message/sipfrag;version=2.0
Supported: 100rel, replaces, norefersub, gruu
Server: Blink 0.8.0 (Windows)
Content-Length: 0

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

<— SIP read from TLS:192.168.1.128:47076 —>
INVITE sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK8140e4b7101BEE68
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]:47076;transport=tls
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_321-UA/4.1.0.84959
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 276

v=0
o=- 1397588446 1397588446 IN IP4 192.168.1.128
s=Polycom IP Phone
c=IN IP4 192.168.1.128
t=0 0
a=sendrecv
m=audio 2226 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
<------------->
— (15 headers 12 lines) —
Sending to 192.168.1.128:47076 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘9002’ for ‘9002’ from 192.168.1.128:47076

<— Reliably Transmitting (NAT) to 192.168.1.128:47076 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK8140e4b7101BEE68;received=192.168.1.128;rport=47076
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone;tag=as2556e460
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(1.8.26.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0a090d3e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:192.168.1.128:47076 —>
ACK sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK8140e4b7101BEE68
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone;tag=as2556e460
CSeq: 1 ACK
Call-ID: [email protected]
Contact: sip:[email protected]:47076;transport=tls
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_321-UA/4.1.0.84959
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

<— SIP read from TLS:192.168.1.128:47076 —>
INVITE sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK1ef3112cBCF798DD
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone
CSeq: 2 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]:47076;transport=tls
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_321-UA/4.1.0.84959
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“9002”, realm=“asterisk”, nonce=“0a090d3e”, uri=“sip:[email protected]:5061;user=phone;transport=tls”, response=“e4e7014e1c2f8e3f6b963af05a6ef312”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 276

v=0
o=- 1397588446 1397588446 IN IP4 192.168.1.128
s=Polycom IP Phone
c=IN IP4 192.168.1.128
t=0 0
a=sendrecv
m=audio 2226 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
<------------->
— (16 headers 12 lines) —
Sending to 192.168.1.128:47076 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘9002’ for ‘9002’ from 192.168.1.128:47076
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101

<— Reliably Transmitting (NAT) to 192.168.1.128:47076 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK1ef3112cBCF798DD;received=192.168.1.128;rport=47076
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone;tag=as2556e460
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.26.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:192.168.1.128:47076 —>
ACK sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.128:47076;branch=z9hG4bK1ef3112cBCF798DD
From: “Ajay” sip:[email protected]:5061;tag=41922E42-87B9F3
To: sip:[email protected];user=phone;tag=as2556e460
CSeq: 2 ACK
Call-ID: [email protected]
Contact: sip:[email protected]:47076;transport=tls
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_321-UA/4.1.0.84959
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
PBX1CLI> sip set debug off
SIP Debugging Disabled
PBX1
CLI>

Making call to 9001 which has TLS and SRTP enabled. Please help

SIP config
tlsenable = yes
tlsbindaddr = 192.168.1.28
tlscertfile = /etc/asterisk/keys/asterisk.pem
tlsdontverifyserver = no
tlscipher=DES-CBC3-SHA
tlsclientmethod = tlsv1
encryption = no

extension 9002
dtmfmode = RFC 2833
trustpid = yes
sendrpid = no
type = friend
nat = yes
port = 5061
qualify = yes
qualify freq 60
transport = TLS only
encryption = Yes (SRTP only)