[asterisk-users] Inbound call not dialing exten

[mahesh katta]

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com



On Thu, Jun 16, 2011 at 1:40 PM, A J Stiles
<asterisk_list at earthshod.co.uk>wrote:

> On Thursday 16 Jun 2011, mahesh katta wrote:
> > Hi all,
> >
> > I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
> > extensions. when incomming call come to this DID no. (4578901) that time
> > 5001 extestinsion should ring.
> > below my dial plan is not getting any result , inthat has any mistakes.
> > please help.
> >
> > exten => _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
> > exten => _45789XX,1,Set(Dest=2{EXTEN:-2})
> > exten =>
> >
> _45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM
> >}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =>
> > _45789XX,3,Dial(SIP/${Dest},,tTo)
> > exten => _45789XX,4,Hangup
>
> Firstly, you've got two "1" steps in that.  Unless you are doing something
> complicated with GOTOs  (and if you are, then there's probably a better way
> of doing it),  use "1" for the first step and "n"  (next)  for all
> subsequent
> steps.
>
> If nothing else, it means you can add extra NoOp() statements to put
> debugging
> information on the console, and later comment out or remove them without
> forced renumbering  (which brings its own opportunities to introduce
> errors).
>
>
> Secondly, you're setting ${Dest} to "2" followed by the last 2 digits of
> the
> dialled number.  But what you really want is "50" followed by the last 2
> digits of the dialled number.  So it should be
>
> exten => _45789XX,n,Set(Dest=50{EXTEN:-2})
>
> >>> sir when I add this its not ringing on sip exten
log will come like
Accepting call from '0559566768' to '4578924' on channel 0/3, span
1

    -- Executing AGI("Zap/3-1", "agi://127.0.0.1:4577/call_log") in new
stack

    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning
0

    -- Executing Set("Zap/3-1", "Dest=50{EXTEN:-2}") in new
stack

    -- Executing MixMonitor("Zap/3-1",
"/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2225.gsm|av(0)V(0)")
in new stack
    -- Executing Dial("Zap/3-1", "SIP/50{EXTEN:-2}||tTo") in new
stack

  == Begin MixMonitor Recording
Zap/3-1

  == Parsing '/etc/asterisk/manager.conf':
Found

  == Manager 'sendcron' logged on from
127.0.0.1

  == Manager 'sendcron' logged off from
127.0.0.1

Jun 16 12:20:07 WARNING[7409]: chan_sip.c:2018 create_addr: No such host:
50{EXTEN

Jun 16 12:20:07 NOTICE[7409]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to
destination)
  == Everyone is busy/congested at this time
(1:0/0/1)

    -- Executing Hangup("Zap/3-1", "") in new
stack

  == Spawn extension (default, 4578924, 5) exited non-zero on
'Zap/3-1'

    -- Executing DeadAGI("Zap/3-1", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------")
in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses

>
> --
> AJS
>
> Answers come *after* questions.
>
> --
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