[asterisk-users] DID is not working (call is not routing)

[Crazy Boy]

Hi,

I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.

When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: 

*CLI>
    -- IAX2/teliax-2 answered SIP/350-09e3b540
    -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk||1") in new stack
    -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack 
    -- Channel will hangup at 2006-10-06 11:27:55 UTC.
    -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack
    -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack 
    -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
    -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack 
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020'
    -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack
    -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack 
    -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack 
    -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2006-10-06 11:28:04 UTC.
    -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack 
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'
I tried in changing the "Features" on my account page. But, no use. Here I am enclosing my configuration files also. Please solve my problem. Looking forward to your response. Thank you.\n

Regards,
Chandra.
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When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully.  Please tell me the solution. Looking forward to your response. Thank you. 

Regards,
Chandra.

 		
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