Your server seems to be doing exactly what you are telling it to do:
 
 -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack
 -- Playing 'ss-noservice' (language 'en')
 
Read the extensions.conf directions on the wiki site:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
 
bp

 
On 10/8/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
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'

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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