Alex is pointing you in the right direction. You should want a single
daemon running that then gets notified by the voicemail script, either
through a FIFO, a socket, or by dropping a file in a watched
directory.
If you are going to write a daemon, I would suggest looking at :
by
the database. what i've noticed is, after the originate, the script never
does anything else. it seems i have to use Async or the AMI will
disconnect, so i tried using OriginateHack=1 but still no dice... any
ideas?
On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote:
Alex
need to see?
On Mon, May 16, 2011 at 2:45 PM, Ryan Bullock rrb3...@gmail.com wrote:
A normal Originate over the AMI will block all other actions until it
completes. So to do other commands while the Originate is still going
you have to call Originate with the Async option. I would suggest
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to test.
Here is an example of what I see on the manager interface during a
register/unregister:
Event: PeerStatus
Privilege:
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
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Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
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Check out the 'p' option for the Dial command.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
It enables call screening, so you have to press 1 to answer. This can also
prevent the voice mail from being left on your cell phone.
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Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
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Catches 555 through 559:
exten = _55[5-9],1,answer
exten = _55[5-9],n,playback(beep)
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
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Ah, sorry, I totally missed that in your description.
Other than the speech recognition that Danny is suggesting, my only thought
is to use an agi that will originate another leg, run AMD (answering machine
detect) and then dump the two parties into a conference to re-join them(or
use the Bridge
Are you running asterisk in a virtual machine?
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So I be it sounds like all the recordings are underwater.
Are you using dahdi for timing? Can you run dahdi_test?
Asterisk needs a good timing source, in the case when you don't have a
physical card providing it, it relies on kernel ticks or the RTC (or HPET).
Because of the nature of virtual
Have you tried 'type = friend', might also want to make sure 'allowguest' is
set to 'no', as this may be putting guest calls into your default context.
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Hey Philipp,
You can check out
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk for
setting up from brute force detection and blocking with asterisk. There are
also a link at the bottom about rate limiting registrations via iptables.
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Please post your results as a note for the issue.
Thanks.
Ryan Bullock
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