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Crazy Boy

2006-Oct-08 07:44 UTC

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

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.
\n\n",0] );  //-->

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.

 		
---------------------------------
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.  Great rates
starting at 1?/min.
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William Piper

2006-Oct-08 19:52 UTC

head link

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

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 <crazymoonboy@yahoo.com>
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.
>
> ------------------------------
> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates
> starting at 1?/min.
>
<http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://messenger.yahoo.com>
>
>
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asterisk users - Oct 2006 - DID is not working (call is not routing)

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

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