Unable to register my soft phone with Astersik 11.15.0

I have isntalled asterisk on the Linux fedora 20. I am a newbie. So need help here. I am unable to register my soft client that is running on another machine. I get the message “Request timeout” on my client. I have checked the “tcpdump” traces on the linux machine and I see following message:

21:11:34.153648 IP (tos 0x0, ttl 128, id 9316, offset 0, flags [none], proto UDP (17), length 685)
192.168.1.6.52924 > 192.168.1.8.sip: SIP, length: 657
REGISTER sip:192.168.1.8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:52924;branch=z9hG4bK-d8754z-dcede1e159b7c197-1—d8754z-
Max-Forwards: 70
Contact: sip:1000@192.168.1.6:52924;rinstance=53a425e0db0a7cf7;transport=UDP;expires=0
To: sip:1000@192.168.1.8;transport=UDP
From: sip:1000@192.168.1.8;transport=UDP;tag=167f2e40
Call-ID: ZmFhNGZhMjczZTcwNzI1NDc4MmU4MjQ5OTg2YTkwODg.
CSeq: 7 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0
21:11:34.158696 IP (tos 0x0, ttl 128, id 9317, offset 0, flags [none], proto UDP (17), length 738)
192.168.1.6.52924 > 192.168.1.8.sip: SIP, length: 710
SUBSCRIBE sip:1000@192.168.1.8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:52924;branch=z9hG4bK-d8754z-26cbb335e9bd2a74-1—d8754z-
Max-Forwards: 70
Contact: sip:1000@192.168.1.6:52924;transport=UDP
To: sip:1000@192.168.1.8;transport=UDP
From: sip:1000@192.168.1.8;transport=UDP;tag=3255a220
Call-ID: MGRiNmFiNGM3NDc3ODY4YzY1ZGMwZGE3MWJjOGFmZDc.
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0


Here is my sip.conf file:
[general]
context=public ; Default context for incoming calls. Defaults to 'default’
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
srvlookup=no ; Disable DNS SRV lookups on outbound calls

;nat=force_rport,comedia ; assume device is behind NAT
; *** NAT stands for Network Address Translation,
; which allows multiple internal devices to share an
; external IP address.

[authentication]
basic-options ; a template
dtmfmode=rfc2833
context=from-office
type=friend
natted-phone ; another template inheriting basic-options
directmedia=no
host=dynamic
public-phone ; another template inheriting basic-options
directmedia=yes
my-codecs ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
ulaw-phone ; and another one for ulaw-only
disallow=all
allow=ulaw

[1000]
type=friend
context=internal
;nat=force_rport,comedia
allow=ulaw,alaw
secret=12345
host=dynamic

[1001]
type=friend
context=internal
;nat=force_rport,comedia
allow=ulaw,alaw
secret=12345
host=dynamic

[1002]
type=friend
context=internal
;nat=force_rport,comedia
allow=ulaw,alaw
secret=54321
host=dynamic

Here is my extensions.conf file:
[internal]

exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)

;exten => 1001,1,NoOp(First Line)
;exten => 1001,2,NoOp(Second Line)
;exten => 1001,3,HangUp


SIP client that I am using : Zoiper ( I have also tried with “Blink” but that is too not working)

What do the Asterisk logs show. In particular, what do they show with “sip set debug on”