Call failed

Hello,
I am trying on 2 PCs, First act as sip server & client (has asterisk & twinkle installed) with username 100 and the other as a client only(has twinkle only installed) with username 101 . I made the configuration and Asterisk successfully runned many times and I made a call between the 2 PCs using Ethernet connection (in abscence of wireless).
I wrote in the extensions.conf:
[globals]
[general]
[internal-phones]
exten => 100,1,Dial(SIP/100,60)
exten => 101,1,Dial(SIP/101,60)
exten =>200,1,Answer()
same=>n,Playback(hello-world)
same=>n,Hangup()

When I try to call extension 101 on twinkle from the twinkle account of 100 , I got “call failed , 404 Not found” & sometimes I got “call unaithorized” but sometimes I can make the call successfully !.. I don’t know why & on what does it depend!!!

When I try to call extension 200 from twinkle , also I got "call failed , 404 not found "!!! I am sure that “hello-world” exits , I don’t know what is the problem .

Note that :registration succedded on Asterisk & Twinkle . Hence , No problem in registration & sip.conf

Here is the ouptut when I call 200 :

<------------>
Scheduling destruction of SIP dialog ‘vjdqmitcnkteiys@sara-Inspiron-N5010’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5062 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bKvuujwlie
Max-Forwards: 70
To: “100” sip:100@192.168.1.2
From: “100” sip:100@192.168.1.2;tag=ezljd
Call-ID: vjdqmitcnkteiys@sara-Inspiron-N5010
CSeq: 160 REGISTER
Authorization: Digest username=“100”,realm=“asterisk”,nonce=“129e1660”,uri=“sip:192.168.1.2”,response=“ff1f2f365327ed1a39eb2810c4022160”,algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.2:5062 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bKvuujwlie;received=192.168.1.2;rport=5062
From: “100” sip:100@192.168.1.2;tag=ezljd
To: “100” sip:100@192.168.1.2;tag=as3862f68f
Call-ID: vjdqmitcnkteiys@sara-Inspiron-N5010
CSeq: 160 REGISTER
Server: Asterisk PBX 1.8.27.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 108
Contact: sip:100@192.168.1.2:5060;expires=108
Date: Fri, 16 May 2014 11:00:31 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘vjdqmitcnkteiys@sara-Inspiron-N5010’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5062 —>
INVITE sip:200@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bKajzmfkul
Max-Forwards: 70
To: sip:200@192.168.1.2
From: “100” sip:100@192.168.1.2;tag=uizdm
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 249 INVITE
Contact: sip:100@192.168.1.2:5062
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 307

v=0
o=twinkle 1339898127 1466682349 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
— (13 headers 14 lines) —
Sending to 192.168.1.2:5062 (no NAT)
Using INVITE request as basis request - sboobqzwuvdkksa@sara-Inspiron-N5010
Found peer ‘100’ for ‘100’ from 192.168.1.2:5062

<— Reliably Transmitting (no NAT) to 192.168.1.2:5062 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bKajzmfkul;received=192.168.1.2;rport=5062
From: “100” sip:100@192.168.1.2;tag=uizdm
To: sip:200@192.168.1.2;tag=as48255104
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 249 INVITE
Server: Asterisk PBX 1.8.27.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="02ccdc5a"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘sboobqzwuvdkksa@sara-Inspiron-N5010’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.2:5062 —>
ACK sip:200@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bKajzmfkul
Max-Forwards: 70
To: sip:200@192.168.1.2;tag=as48255104
From: “100” sip:100@192.168.1.2;tag=uizdm
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 249 ACK
User-Agent: Twinkle/1.4.2
Content-Length: 0

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

<— SIP read from UDP:192.168.1.2:5062 —>
INVITE sip:200@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bKjphtbbvv
Max-Forwards: 70
To: sip:200@192.168.1.2
From: “100” sip:100@192.168.1.2;tag=uizdm
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 250 INVITE
Contact: sip:100@192.168.1.2:5062
Content-Type: application/sdp
Authorization: Digest username=“100”,realm=“asterisk”,nonce=“02ccdc5a”,uri="sip:200@192.168.1.2",response=“b8e7643d1495918f78e03385034302af”,algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 307

v=0
o=twinkle 1339898127 1466682349 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
— (14 headers 14 lines) —
Sending to 192.168.1.2:5062 (no NAT)
Using INVITE request as basis request - sboobqzwuvdkksa@sara-Inspiron-N5010
Found peer ‘100’ for ‘100’ from 192.168.1.2:5062
== Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format speex for ID 98
Found audio description format speex for ID 97
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x20000020e (gsm|ulaw|alaw|speex|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:8000
Looking for 200 in internal-phones (domain 192.168.1.2)

<— Reliably Transmitting (no NAT) to 192.168.1.2:5062 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bKjphtbbvv;received=192.168.1.2;rport=5062
From: “100” sip:100@192.168.1.2;tag=uizdm
To: sip:200@192.168.1.2;tag=as48255104
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 250 INVITE
Server: Asterisk PBX 1.8.27.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘sboobqzwuvdkksa@sara-Inspiron-N5010’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.2:5062 —>
ACK sip:200@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bKjphtbbvv
Max-Forwards: 70
To: sip:200@192.168.1.2;tag=as48255104
From: “100” sip:100@192.168.1.2;tag=uizdm
Call-ID: sboobqzwuvdkksa@sara-Inspiron-N5010
CSeq: 250 ACK
Authorization: Digest username=“100”,realm=“asterisk”,nonce=“02ccdc5a”,uri="sip:200@192.168.1.2",response=“b8e7643d1495918f78e03385034302af”,algorithm=MD5
User-Agent: Twinkle/1.4.2
Content-Length: 0

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