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<br><br><div class="gmail_quote">On Thu, Jun 16, 2011 at 1:40 PM, A J Stiles <span dir="ltr"><<a href="mailto:asterisk_list@earthshod.co.uk">asterisk_list@earthshod.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im">On Thursday 16 Jun 2011, mahesh katta wrote:<br>
> Hi all,<br>
><br>
> I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099<br>
> extensions. when incomming call come to this DID no. (4578901) that time<br>
> 5001 extestinsion should ring.<br>
> below my dial plan is not getting any result , inthat has any mistakes.<br>
> please help.<br>
><br>
> exten => _45789XX,1,AGI(agi://<a href="http://127.0.0.1:4577/call_log" target="_blank">127.0.0.1:4577/call_log</a>)<br>
> exten => _45789XX,1,Set(Dest=2{EXTEN:-2})<br>
> exten =><br>
> _45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM<br>
>}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =><br>
> _45789XX,3,Dial(SIP/${Dest},,tTo)<br>
> exten => _45789XX,4,Hangup<br>
<br>
</div>Firstly, you've got two "1" steps in that. Unless you are doing something<br>
complicated with GOTOs (and if you are, then there's probably a better way<br>
of doing it), use "1" for the first step and "n" (next) for all subsequent<br>
steps.<br>
<br>
If nothing else, it means you can add extra NoOp() statements to put debugging<br>
information on the console, and later comment out or remove them without<br>
forced renumbering (which brings its own opportunities to introduce errors).<br>
<br>
<br>
Secondly, you're setting ${Dest} to "2" followed by the last 2 digits of the<br>
dialled number. But what you really want is "50" followed by the last 2<br>
digits of the dialled number. So it should be<br>
<br>
exten => _45789XX,n,Set(Dest=50{EXTEN:-2})<br>
<font color="#888888"><br></font></blockquote><div>>>> sir when I add this its not ringing on sip exten <br>log will come like <br>Accepting call from '0559566768' to '4578924' on channel 0/3, span 1 <br>
-- Executing AGI("Zap/3-1", "agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a>") in new stack <br>
-- AGI Script agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a> completed, returning 0 <br> -- Executing Set("Zap/3-1", "Dest=50{EXTEN:-2}") in new stack <br>
-- Executing MixMonitor("Zap/3-1", "/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2225.gsm|av(0)V(0)") in new stack <br><span style="color: rgb(255, 0, 0);"> -- Executing Dial("Zap/3-1", "SIP/50{EXTEN:-2}||tTo") in new stack </span><br>
== Begin MixMonitor Recording Zap/3-1 <br> == Parsing '/etc/asterisk/manager.conf': Found <br>
== Manager 'sendcron' logged on from 127.0.0.1 <br> == Manager 'sendcron' logged off from 127.0.0.1 <br>
Jun 16 12:20:07 WARNING[7409]: chan_sip.c:2018 create_addr: No such host: 50{EXTEN <br>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) <br>
== Everyone is busy/congested at this time (1:0/0/1) <br> -- Executing Hangup("Zap/3-1", "") in new stack <br>
== Spawn extension (default, 4578924, 5) exited non-zero on 'Zap/3-1' <br> -- Executing DeadAGI("Zap/3-1", "agi://<a href="http://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------">127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------</a>") in new stack <br>
-- AGI Script agi://<a href="http://127.0.0.1:4577/call_log--HVcauses">127.0.0.1:4577/call_log--HVcauses</a> <br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<font color="#888888">
<br>
--<br>
AJS<br>
<br>
Answers come *after* questions.<br>
</font><div><div></div><div class="h5"><br>
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