I know that on most PBXs, you can wire an FXO port to overhead paging speakers while supplying talk-battery in order to have overhead paging without any specialized hardware setup. A Valcom VP-324 power supply is ample to supply the talk-battery. You just wire the trunk in question in sequence with the power supply and the overhead speakers.

Is Asterisk capable of picking up on a "dead" trunk in order to do something similar to this? My guess would be to create an outbound path to dial "nothing" on whichever FXO port is used, but is there a way to make Asterisk pick up the FXO port even though there's technically no line there?

Undrhil

The above link in the article doesn't exist anymore, so here is a copy from Google cache :

#!/usr/bin/perl

#
# allpage.agi - Copyright Rob Thomas (xrobau@gmail.com) 2005.
#
# Revision 1.1 - 14th October 2005 - Added Polycom Support
#
# This program is free software; you can redistribute it and/or
# modify it under the terms of Version 2 of the GNU General
# Public License as published by the Free Software Foundation
#
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
# Simple AGI to page all SIP extensions (no IAX device, because at the time
# of writing this, no device supported IAX_ANSWER_IMMED) that aren't on
# the phone. Tested with Asterisk 1.2. Should work out-of-the-box with
# Grandstream GXP phones with firmware greater than 1.0.12, and Snoms with
# 'enable intercom' on and 'filter packets from registrar' off.
#
# Documentation:
# Your dialplan consists of two things. Firstly, in the context that your
# normal phones are in, you need to have something like this:
# exten => 999,1,AGI,allpage.agi
# exten => 999,2,MeetMe(999,dq)
# exten => 999,3,Playback(beep)
# exten => 999,4,Hangup
#
# The paged phones then jump to this context:
# all-page
# exten => s,1,AbsoluteTimeout(10)
# exten => s,2,MeetMe(999,dmq)
# exten => s,3,Hangup
# exten => t,1,Hangup
# exten => T,1,Hangup
#
# Any questions? Join
#openpbx on irc.freenode.net
# --Rob Thomas 28th Sep, 2005.
if (eval "require Net::Telnet;")
{ use Net::Telnet; }
else { print "VERBOSE \"Net::Telnet NOT INSTALLED - this is required\" 0\n"; exit 0; }
# You need to configure this: Your manager API username and password. This
# is the information from /etc/asterisk/manager.conf. You need something like
# this in it:
# admin
# secret = amp111
# deny=0.0.0.0/0.0.0.0
# permit=127.0.0.0/255.0.0.0
# read = system,call,log,verbose,command,agent,user
# write = system,call,log,verbose,command,agent,user
# IF that's what you have in your conf file, this is what you should have here:
my $mgruser = "admin";
my $mgrpass = "amp111";
my $mgrport = 5038;
# If you're using a SNOM they need a 'sip:ip.add.re.ss' added to the Call-Info field,
# with the IP address of the registrar. Most other phones will silently ignore this,
# but if you have trouble, you may need to fiddle with this line. Change the IP Address
# to be that of your Asterisk Server.
my $callinfo = 'Call-Info: sip:192.168.0.1\; answer-after=0';
# This is for Polycom phones - see
# http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config
my $alertinfo = 'Ring Answer';
# That's it. Nothing else should need to be changed
# Some variables we use later...
my @tocall;
my %useref;
# Get all SIP channels:
my @allsip = `asterisk -rx "sip show peers"`;
# Get all that are in use:
my @inuse = `asterisk -rx "sip show channels"`;
# Strip off carrige returns and take off the top line
chomp @allsip; chomp @inuse; shift @allsip; shift @inuse;
# First, we want to get the phones that are in use. We need the second field
# of every line.
while (my $line = shift @inuse) { my @tmparr = split(/\s+/,$line);
# The second last, or possibly last, line says 'n active SIP channel(s)'
goto endinuse if ($tmparr1 eq "active"); print "VERBOSE \"Found extension $tmparr1 in use.\" 1\n"; $useref{$tmparr1} = "In Use"; } endinuse:
# We only want the first column, and, we only want the first part before the
# slash
while (my $line = shift @allsip) {
# This may break if there's more than one / in a line. There shouldn't.
$line =~ /(.+)\/.+/;
# Sanity check. This may be a peer for outgoing calls, rather than an
# extension. Basically, if it's 5 numbers or less, it's an extension.
my $result = $1;
# If your dialplan is different, sucks to be you. Change this regexp to
# match your dialplans and EXCLUDE your SIP trunks.
if ($result =~ /^\d{1,5}$/) { if (defined $useref{$result}) { print "VERBOSE \"NOT Adding extension $result to call list\" 2\n"; } else { print "VERBOSE \"Adding extension $result to call list\" 1\n"; push @tocall, $result; } } }
# If you don't want any intelligence, you can just delete all the logic above
# here, and specify the SIP extensions to call here. Also useful for debugging.
#@tocall = (303, 301);
# Now, we have an array (@tocall) with all valid SIP extensions.
while (my $sipxtn = shift @tocall) { print "VERBOSE \"Doing $sipxtn\" 0\n";
# Open connection to AGI
my $tn = new Net::Telnet ( Port => $mgrport, Prompt => '/.*$%#> $/', Output_record_separator => '', Input_Log=> "/tmp/input.log", Output_Log=> "/tmp/output.log", Errmode => 'return', );
$tn->open("127.0.0.1");
$tn->waitfor('/0\n$/');
$tn->print("Action: Login\n");
$tn->print("Username: $mgruser\n");
$tn->print("Secret: $mgrpass\n");
$tn->print("Events: off\n\n");
my ($pm, $m) = $tn->waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) { print "VERBOSE \"Incorrect MGRUSER or MGRPASS - unable to connect to manager interface\" 0\n";
exit;
}
$tn->print("Action: Originate\nChannel: SIP/$sipxtn\nContext: all-page\nPriority: 1\n");
$tn->print("Variable: SIPADDHEADER=\"$callinfo\"\n");
$tn->print("Variable: ALERT_INFO=\"$alertinfo\"\n");
$tn->print("Extension: s\nCallerID: System Page\n\n");
$tn->print("Action: Logoff\n\n");
$tn->close;
}

Has anyone successfully set up a polycom to auto-answer with Broadsoft?