Please note that while this system is still in operation for now the services it's providing need to be offloaded to somewhere (and someone) else if it is to be around for more than another couple of years. We've begun the process to move to a more transient lifestyle within the next few years at which point I will be relocating my basic internet services (email, websites) to a co-located or virtual server and will be eliminating enhanced services such as this PBX and my audio/video streams. Currently this system is also handling our homes telephones as well so I'm not in a position to hand over the TDM hardware just yet. However, I would like to begin transition to someone else in the group so the great idea of (nearly) free telephony can continue to grow. If you have time to contribute to the group, are experienced with operating Linux servers, can pickup the basics of the Asterisk PBX and would like to explore the idea of taking this thing over please send an email to nat@personaltelco.net. I promise that I will give ample warning (a few months) to the list before the system is actually taken off line if no workable solution pops up. - Nat

PersonalTelcoPbx is a telephone system providing user friendly voice communications that very closely resembles what you get with your standard telephone company but with more features and less (nearly zero) cost. It currently consists of a single Linux host running the Asterisk open source PBX software with some special telephony hardware inside but will hopefully expand to a network of similar servers.

External Access

• Gateways currently exist to the public switched telephone network (PSTN), Free World Dialup, IAXTel, e164.org, [http://www.voipuser.org/ VoIP User] and of course SIP on the Internet. Plans are in the works for integration with ENUM and e164.org but these have not been fully implemented yet. Here are the numbers you can use to reach the PtpPbx from these telephone networks:

  PSTN (like Qwest or Verizon)	(503) 914-6550
  Free World Dialup	274185
  IAXTel	(700) 914-6550
  SIP	sip://5039146550 @sip.personaltelco.net
  e164.org (International ENUM)	88299 000079 00

Dialing Plans

• Depending on your context you may have different dialing options available. Currently there are two visible contexts: incoming caller and registered user. The incoming caller context is what you get when you reach the PBX by dialing one of the numbers listed in the External Access section, the registered user context is what's available on phones (hardware or software) that are attached to a registered account on the PBX. To dial a registered PBX user enter their three digit extension any time during the opening etiquette. For example, if you're trying to reach extension 112 simply dial 112 (not 1112).

Incoming Callers

• Option	Description
  0	Directory of PTP Extensions
  1+NN	Dial PTP Extension (extensions are three digits long including the one)
  4	Tri-met Bus Arrival Utility
  6	General Conference Room (Public)
  *	Voicemail Administration

Registered Users

• Option	Description
  0	Directory of PTP Extensions
  1+NN	Dial PTP Extension (extensions are three digits long including the one)
  3	Echo Test
  4	Tri-met Bus Arrival Utility
  6	General Conference Room (Public)
  7+NNNNNN	Call Free World Dialup User
  88299+NNNNNN NN	Call e164 international number
  9+NNN NNN NNNN	Call 10 digit number on the PSTN (Free to USA and Canada)
  *	Voicemail Administration

Getting an Account

Until a solution can be worked out to move operation of this PBX to someone else no new accounts will be created. - Nat

Wish List

1. Administration Tool - CGI scripts that allow easy maintenance of accounts, extensions and voicemail boxes. Currently this sort of stuff is modified by editing the Asterisk config files by hand and executing reload against the server.

2. Voice Messages put in a Web getable directory/page

3. Queue up outbound calls if no lines are available - If an internal user attempts to call out to PSTN and none of those lines are available it would be nice if the caller had the option to stay on the line until one is open or have the system ring them back when their call can be routed.

4. FRS <-> Conference Room Gateway - A gateway that would allow users in the general conference room to hear traffic on FRS channel 11 in Portland and possibly request a broadcast by hitting pound to start talking and pound again when finished talking.

Log of Granted Wishes and Other Changes

• CDR database in MySQL - 05/08/2004 - I've compiled in the mysql_cdr addon and now have call details logging to a mysql table. This will eventually be accessed by a web-based tool for users to show their telephone usage.

• All lines busy message - 05/07/2004 - I've recorded a greeting on the Vonage voicemail system at 503-914-6550 that states something like "all incoming telephone lines are currently busy, try back in a few minutes" that will be heard if someone calls the single incoming PSTN line when it's already in use. If an outbound call is attempted that requires a PSTN line and none are available a message is played explaining something like "all outbound public telephone network lines are currently unavailable, please try again in a few minutes."

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