I've tried every script and suggestion on this site, but I still have a problem with paging approx. 50 Aastra 9133i phones. The problem is that it can take up to 20 seconds for all of the phones to connect to the page. Some phones connect right away, while others seem to take forever. Anyone else run into this problem? Any solutions or suggestions on how to solve this?

Somebody use VPN to solve the VoIP Blocking issue. But it seems not a good way to solve the voip blocking issue. Because VPN will take more bandwidth and will take effection on the Voice Quality

Currently I am using the VGCP, a new solution to solve the VoIP Blocking issue. Following is theirs website:
http://www.speed-voip.com/index-36.html

If any of you have interested, you may try to use it to solve your VoIP Blocking problems. Thanks.

Andy
andywong-01@hotmail.com

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?