[asterisk-users] Sip phone does not call

[ayodele abejide]

Hello group,

 

I have asterisk running on my ubuntu machine, and I have a
peer to peer network with an XP machine, both of the running x-lite client, I try
calling either of the soft phone from the other and the response I get is on my
asterisk console is as below: 

 

 

[May 19 19:31:18] NOTICE[1476]: chan_sip.c:20006
handle_request_invite: Call from '1000' to extension '3000' rejected because
extension not found.

 

[May 19 19:31:39] NOTICE[1476]: chan_sip.c:21298
handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000

 

[May 19 19:31:44] NOTICE[1476]: chan_sip.c:20006
handle_request_invite: Call from '1000' to extension '1000' rejected because
extension not found.

 

 

My Diaplan Settings (extensions.conf)

 

[globals]



 

 

[general]



autofallthrough=yes



 

 

[default]



exten => s,1,Verbose(1|Unrouted call handler)



exten => s,n,Answer()



exten => s,n,Wait(1)



exten => s,n,Playback(tt-weasels)



exten => s,n,Hangup()



 

 

[incoming_calls]



 

 

[internal]



exten => 1000,1,Verbose(1|Extension 1000)



exten => 1000,n,Dial(SIP/1000,30)



exten => 1000,n,Hangup()



 

 

exten => 3000,1,Verbose(1|Extension 3000)



exten => 3000,n,Dial(SIP/1000,30)



exten => 3000,n,Hangup()

 

 

Sip Settings (sip.conf)

 

[general]

context=default

bindport=5060

srvlookup=yes

 

[1000]

type=friend

host=dynamic

context=phones

 

[3000]

type=friend

host=dynamic

context=phones

 		 	   		  
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