[asterisk-users] How to continue processing a context after a Hangup
Satish Barot
satish4asterisk at gmail.com
Fri Jun 3 00:11:53 CDT 2011
Warren,
A good example given.
Just suggest to use 'Move' instead of 'Copy' for placing callfile in
outgoing folder.
A J Stiles has explained it in a better way in one of his replies.
http://lists.digium.com/pipermail/asterisk-users/2011-May/262929.html
[SATISH]
On Fri, Jun 3, 2011 at 1:16 AM, Warren Selby <wcselby at selbytech.com> wrote:
> 2011/6/2 Antonio Modesto <modesto at isimples.com.br>
>
>> Good afternoon,
>>
>> I'm trying to write a simple callback context, but i need to hangup an
>> incoming call and then call the origin number back, the problem is that
>> asterisk stops processing the call after Hangup() application then it is not
>> able to dial the origin number back.
>>
>>
> The way I did it was to use a DeadAGI from the 'h' exten that created a
> call file. This is how I did it for a client on Asterisk 1.4.x:
>
> [rec-call-back-in]
> exten => new,1,Answer()
> exten => new,n,Wait(1)
> exten => new,n,Playback(vm-intro)
> exten => new,n,Playback(beep)
> exten => new,n,Set(timestamp=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
> exten => new,n,Set(FILENAME=reccallback/${CALLERID(num)}-${timestamp})
> exten => new,n,Record(${FILENAME}.gsm,,,q)
> exten => new,n,Playback(vm-goodbye)
> exten => new,n,Hangup()
>
> exten => h,1,Verbose(Hangup after recording)
> exten => h,n,DeadAGI(reccallback.agi,${FILENAME},${TIMESTAMP})
>
> [rec-call-back-out]
> exten => out,1,Wait(2)
> exten => out,n,Playback(${playbackfile})
> exten => out,n,Hangup()
>
> reccallback.agi:
> #!/usr/bin/perl
>
> use Asterisk::AGI;
> use File::Copy;
>
> $AGI = new Asterisk::AGI;
>
> my %input = $AGI->ReadParse();
> my $callerid = $input{'callerid'};
>
> my $recfile = $ARGV[0];
> my $timestamp = $ARGV[1];
>
> open CALLFILE, ">/var/spool/asterisk/tmp/$callerid-$timestamp.call";
> if (length($callerid) > 4) {
> print CALLFILE "Channel: SIP/external-sip-provider/+1$callerid\n";
> } else {
> print CALLFILE "Channel: SIP/$callerid\n";
> }
> print CALLFILE "CallerID: \"CUSTOMER\" <XXXXXXXXXX>\n";
> print CALLFILE "MaxRetries: 2\n";
> print CALLFILE "RetryTime: 60\n";
> print CALLFILE "WaitTime: 20\n";
> print CALLFILE "Context: rec-call-back-out\n";
> print CALLFILE "Extension: out\n";
> print CALLFILE "Priority: 1\n";
> print CALLFILE "Set: playbackfile=$recfile\n";
> close CALLFILE;
> sleep(5);
>
> copy("/var/spool/asterisk/tmp/$callerid-$timestamp.call",
> "/var/spool/asterisk/outgoing/$callerid-$timestamp.call") or die "copy
> failed: $!";
>
> exit;
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> Our website just got a facelift! Check it out!
> http://www.SelbyTech.com <http://www.selbytech.com>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110603/dde5ba6a/attachment.htm>
More information about the asterisk-users
mailing list