Question : asterisk calling problem

Hello experts,

I have installed an asterisk in a debian lenny distro. and now i am trying to place calls between two sip phones(i am using xLite)  installed on two different pc(note these PCs are in a LAN while server is situated remotely) i have made two configuration files sip.conf and extensions.conf

but i am unable to get these two sip hones registered to asterisk server

Further details are as follows:
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sip show peers:
Name/username              Host            Dyn Nat ACL Port     Status           
2000                       (Unspecified)    D          0        Unmonitored      
1000/1000                  (Unspecified)    D          0        Unmonitored      
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
 
sip show settings:
Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  Always auth rejects:    No
  Call limit peers only:  No
  Direct RTP setup:       No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             none
  IP ToS RTP audio:       none
  IP ToS RTP video:       none
  T38 fax pt UDPTL:       No
  RFC2833 Compensation:   No
  SIP realtime:           Disabled
 
Global Signalling Settings:
---------------------------
  Codecs:                 0x10e (gsm|ulaw|alaw|g729)
  Codec Order:            none
  T1 minimum:             100
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
 
Default Settings:
-----------------
  Context:                default
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               (Defaults to English)
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk
 
 
sip show peer 1000:
 
  * Name       : 1000
  Secret       : 
  MD5Secret    : 
  Context      : phones
  Subscr.Cont. : 
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 122.168.224.211
  Addr->IP     : (Unspecified) Port 0
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 1000
  SIP Options  : (none)
  Codecs       : 0x10e (gsm|ulaw|alaw|g729)
  Codec Order  : (none)
  Auto-Framing:  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :
 
sip show domains:
SIP Domain support not enabled.
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Answer : asterisk calling problem

checkout how to setup remote extension on the link below or google it.

http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension


Good Luck!
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