I have looked at many Asteisk packages and AdminsParadise's implementation was one.  I was interested in their cluster setup and how well it worked.

AdminParadise's implementation does not work well at all.  Let me begin by saying Clustering is VERY possible with Asterisk and CentOS 5 is the easiest I found to do it with. 

First they copied the exact code presented by salahuddin66 Dhaka, Bangladesh in 2006. His code is linked to in my links section. 

These are the problems I found with their implementation:

Their install failed to properly copy a file called amportalhb.  This file is nothing more than an old way to make amportal respond to a start and stop command with results which it now does so only a symbolic link is necessary to the amportal file.  All files that must be executed upon a heartbeat fail must respond to a start and stop command - which is why this file was made back in the days when amportal was not using the correct syntax.

Lets get into the real problems - select more to read the remainder of this article...

I started to code a few routines to fix the MWI notice on the phones for my clients.  I was having a real hard time finding any documentation about turning that little light on without digging into a complex SIP protocol manual.

This problem seems to be fixed on the newer version of Asterisk, but not all clients are on the same versions.  I noticed on my newer install that about every 5 to 6 minutes a pass is made and the MWI indicator is set or reset as needed.  This is great for the newer installs, but what about the older stable ones that sometimes don’t get the MWI notice?

I’m finally releasing three routines that should take care of any of your MWI problems.  The three programs are to set the MWI by extension number, reset the MWI by extension number and fix all phones that have voicemail accounts (Set or Reset based on voicemail). 

I like to have links with my client’s machines, but I found no real information on how to do this by SIP.  There are a lot of scraps everywhere, but Asterisk to Asterisk is constantly referenced and described using an IAX link.  If you think about various suppliers like Stanaphone (one of my favorites) linking to their server with yours is documented.  This is a SIP server to SIP server, but where on VOIPinfo is the configuration for both sides?

Overview:

I've read and discussed various methods to track my clients T1 status.  I don't like to reinvent the wheel but sometimes it is necessary.  Using Sangoma's T1 (A101 & A102) cards, but any card would work, I came to realize that most of the T1 outages did not stop the server or Asterisk.

This is not entirely true.  There are cases where Asterisk crashes from an oscillating T1 line, but I noticed there is a usually a lag - enough to send out warnings.  Warnings by email, txt message, beeper and phone calls using the backup phone lines and the email system.

So realizing we have a minute or two before the possible final crash, at best no T1, I wrote a few routines to notify me and the client so the proper steps can be taken.

It goes like this.  A cron job is run every minute or two checking the PRI status (I use PRI interfaces). If the PRI is down a script is run to email the customer instructions on who to contact, the account, circuit and which line to do an RCF (Remote Call Forward) on.  In addition, a call is made to my office, my cell, the clients primary operator extension and a beeper.

There is a failsafe built in to stop too many notifications by using the Asterisk database to hold variables. These variables are set and cleared as the T1 line status changes.  I do not want to be notified more than three times that the T1 is still down and I want to know when it comes back up. 


Lets continue on to the coding....

This will be my new site for Asterisk tips and information.  Please stay tuned, I have to consolidate many different sources to this site.

Due to the nature of voip-info.org my work is splattered with advertisments, modifications and the like.  Newgroups and help list will also be added to this site.  Any emails and software donations I made will be consolidated to bring a new source without being 'claimed' 'stamped' or altered in any way.

Paul Norris

The Zoom 5801 answers a market need for ATA 911 compliance - to a degree. This ATA has a POTS (FXO) connection in addition to the phone connection (FXS). This POTS connection can be engaged by the dialplan set in the ATA, a loss of the VOIP server as well as dialing a special code. So if I deploy these, based on the config file in the ATA, I can set the 911 to use the POTS line, the VOIP line or the pots as a backup to the voip line and I believe the VOIP line in backup to the POTS line for emergency numbers.

Having an FXO port makes you want to use it for Asterisk! Well it can to a point. You can have the ATA act as a single line SIP FXO port (in and out) or an ATA, but not concurrently - at this time. Read on.