In my house i am using an autodialer to dial 74949000 to access to gateway and then i dial my mobile or local number to benefit from the saving Can we do that in asterisk to autodial to the gateway 74949000 and wait 10 before i eneter my destination number Please advise me on that how to make a effective dialplan...suing the above information Thanking you in advance
You can do a couple of things. If you are using a SIP device, or a device that
uses a PLAR code (Private Line Automatic Ringdown) then you can put that
number that you want to dial when the phone goes off-hook in that section.
That way, when you pick up the phone it will dial that number first which
will then prompt you via some sort of IVR. Or, you could create an AGI script
(observe below) to Dial the number, wait ten seconds, then Dial the dialed
digits when you receive confirmation.
--script--
extensions.conf
----
[default]
exten => _.,1,agi,savings.agi|${EXTEN}
exten => _.,2,Hangup
/var/lib/asterisk/agi-bin/savings.agi # Set chmod 755 in order to execute
----
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$AGI->setcallback(\&mycallback);
print STDERR "AGI Environment Dump:\n";
foreach $i (sort keys %input) {
print STDERR "-- $i = $input{$i}\n";
}
my $userid = $input{'calleridname'};
my $exten = $input{'extension'};
my $savings_number = "74949000";
if($exten eq 'h') { exit; }
$AGI->exec("Dial", "Zap/g1/$savings_number"); # Replace
with way that you are
dialing out to PSTN
sleep(10000);
--end script--
On Thursday 18 November 2004 06:16 am, Simon wrote:> In my house i am using an autodialer to dial 74949000 to access to
> gateway and then i dial my mobile or local number to benefit from the
> saving Can we do that in asterisk to autodial to the gateway 74949000 and
> wait 10 before i eneter my destination number
>
> Please advise me on that how to make a effective dialplan...suing the
> above information
>
> Thanking you in advance
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Brian Wilkins
Software Engineer
brian@hcc.net
Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
On Wed, 2006-02-01 at 21:58 -0800, Abhishek wrote:> Channel: SIP/914084704700@sip_proxy-out> even when i can dial out manually through the same context(sip_proxy-out) > in sip.conf.as used in the call file sip_proxy-out shouldnt be a context, it should instead be an account defined in sip.conf. Not sure if you misspoke or not ... If you want to dial via a context try: Channel: Local/12345@context For others looking at this, keep in mind though that it will connect to the Channel first then once connected it will connect to the context or application specified elsewhere in the call file. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060131/1b68fbc5/attachment.pgp
Hi list,
I am facing a problem in auto diailing through call files. When i try to
dial having this in my test.call ,
Channel: SIP/914084704700@sip_proxy-out
Callerid: 3301
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: test_in
Extension: 1235
Priority: 1
I receive :
*CLI> -- Attempting call on SIP/914084704700@sip_proxy-out for
1235@test_in:1 (Retry 1)
> Channel SIP/sip_proxy-out-1ebf was never answered.
Feb 1 10:06:21 WARNING[6019]: cdr.c:548 ast_cdr_disposition: Cause not
handled
Feb 1 10:06:21 NOTICE[6019]: pbx_spool.c:266 attempt_thread: Call failed to
go through, reason 8
even when i can dial out manually through the same context(sip_proxy-out)
in sip.conf.
Can somebody help me out of this problem.
Thanks
Abhishek