DIY Emergency Cell Tower

Chris Paget, KJ6GCG has over a decade of experience as an information security consultant and technical trainer for a wide range of financial, online, and software companies. Chris' work is increasingly hardware-focused, recently covering technologies such as GSM and RFID at venues such as Defcon and Shmoocon.

At the recent Defcon 18 conference he displayed a spoofed GSM cellphone tower using a Universal Software Radio Peripheral (USRP) transmitting 25 milliwatts, to present a GSM air interface to a standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in the developing world.

http://openbts.sourceforge.net/

FCC rules aside, this is some powerful stuff. Imagine being able to restore cellular coverage in a disaster area using the foundation he has laid.

You can read more here:

http://www.networkworld.com/news/2010/083010-open-source-voip-cell-phones-at-burning-man.html

Here is a video that shows a test call being placed from a softphone talking to an Asterisk PBX in conjunction with GNU radio and a USRP to create a Part 15 signal level call to a cellphone.




Regarding using HSMM style techniques for running an amateur cell site:

http://www.innismir.net/article/513


Here is a similar project which modifies android phone to use asterisk running on something called a mesh potato. This would be great for developing areas. Just drop some of the MP's with a battery and a solar cell and poof... a phone system.

http://www.villagetelco.org/

Another favorite is using asterisk with amateur radio and repeaters

http://ohnosec.org/drupal/

HSMM VOIP Network in Spain

There is an interesting HSMM/ Asterisk VOIP network in Spain:

http://translate.google.com/translate?hl=en&sl=es&u=http://www.bicubik.net/hsmmn/

Alex EA5HJX, Andres EA5HIQ, Pepe EA5SW, Paco EB5HTC, Javier EB5BXA, Boletin EA5SW, Paco EB5EA, Sergio EA5HFB, Ernesto EB5JDY and several other hams are researching new technologies and telecommunication systems.

Can you explain are in street language, What is Intended and this project will bring to ham radio and its benefits?

So we can all understand, a HSMM network is just one of many highway lanes that can move large amounts of data at high speed. What would be technologically wireless broadband network of high capacity in her ability to integrate all technologies: RF, VoIP, multimedia, data ....

What are the Objectives of the Project HSMMN?

To promote knowledge and new technologies to the general public: The HSMM project has not only a technical aspect but also a social aspect, with aims to bring new technologies and their application in the real world to the public with expertise or not. Disseminating knowledge and new technologies to the general public: The project has HSMMN Not Only But Also to Technical aspect of social aspect, with aims to bring new technologies and their application in the real world to the public with expertise or without them.

We intend that anyone interested can collaborate, learn and practice the knowledge acquired. Anyone interested can collaborate, learn and practice the acquired knowledge. To this end, talks and workshops are planned in various radio clubs and associations interested in the project. To this end, talks and workshops are planned in various radio clubs and Associations interested in the project.
    
Creation of a research group developing new technologies to the world of amateur radio: This project not only aims to develop the network, which is the chief end of it, but to bring together people interested in researching and learning about new technologies to help amateur radio operators and emergency services communications.

Unifcar technologies: using computer systems and hardware elements to create an IP data highway you can travel by all types of information: audio, video, telemetry, APRS, etc ...

Collecting, processing and sending information through a variety of different technologies such as for sending APRS weather information (winds, rainfall, air pressure, barometric pressure, etc)

Ability to provide automated voice announcements from a centralized control room radio repeaters as required. Capacity to Provide automatic voice announcements from a centralized control room radio repeaters as required.

Creating an information service of communications systems available by region, by using a geographic number to gain access from the fixed, mobile and IP (VoIP).

Integration of other communication networks: IRLP, DSTAR, WIRES-II, E-QSO, etc.

Interconnection of repeaters using wireless technologies: Wireless, WiMAX. Each repeater could put a computer capable of processing all the telephony and transform it into Voip (voice over IP) that is transported through the network to reach another repeater HSMMN and is decoded by joining together all repeaters creating a mesh. Each repeater Could put a computer capable of processing all the telephony and transform it into VoIP (Voice over IP). That Is Transported through the HSMM network Reaches Another repeater network and is decoded by joining together all repeaters creating a mesh.

The whole process will be use Free Software.


Entities in the world of emergencies communications in Spain have shown interest in this project as an additional operational tool to consider for future medium to long term. Access from the PSTN will be in collaboration with guifi.net. This is an area open, free and neutral telecommunications network built through a peer to peer agreement where everyone can join the network

It looks like a nice mix of HSMM, Asterisk SIP and app_rpt technologies.

Alex Casanova EA5HJX, writes that he is working to develop a box that is a router (IP), and Asterisk (VoIP) and gateway with radio (analog and digital radio). Maybe this box can help in disaster like Haiti. Around the HSMMN´s Project we are investigate to integrate Asterisk VoIP with analog or digital radios.

NWR SAME software decoder?

Server weather season is upon us.

I have often thought it would be nice if there was an open source (soundcard/ FOB based) SAME decoder solution.

One could dedicate a cheap USB sound FOB to a receiver parked on their NOAA weather radio frequency that would sit and decode any SAME data bursts.

I am thinking for interfacing to repeaters to provide custom weather alert signaling.

It does appear that software to decode SAME data exists, just not open source.

http://www.dxsoft.com/en/products/seatty/

A SAME software decoder would benefit projects like thelinkbox, and asterisk app_rpt as well as other projects.

{Update 6/11}
Greg Hewgill, has updated the source to his NWR tools, now at:

https://github.com/ghewgill/nwr


"Drew" Kirkman, W4KMC writes:
"TECHNICAL INFORMATION:
NOAA’s Specific Area Message Encoding (or SAME) protocol is used to further streamline the Emergency Alert System. Information about an emergency message (such as locations affected, type of message, where it’s coming from, and how long it will be considered effective) is transmitted in the form of digital bursts at the beginning and end of said message. These bursts are AFSK-modulated data with a throughput of 520.83 bits per second. Mark tone (binary 1) is 2083.3 Hz and space tone (binary 0) is 1562.5 Hz, with each tone lasting about 1920 microseconds. Bytes are transmitted in reverse order (LSB -> MSB), that is, 00010111 would be transmitted as 11101000. There are other technical specifications regarding its use in the real world, but it’s irrelevant here. Essentially, if you handed the right text to it, I have a SAME encoder. It outputs true SAME-encoded data."

See his NOAA SAME web based audio encoder/decoder at:
http://www.drewkirhttp://www.blogger.com/img/blank.gifkman.com/projects/noaa-same/

I also stumbled into:

"Using an Arduino Uno and a few other external components, I've been able to reliably decode the SAME messages."

http://www.raydees.com/Weather_Radio.html

{Update 2012}
Someone updated multimon, and it now has EAS / SAME decoding support!
https://github.com/EliasOenal/multimonNG/blob/master/README

And this PHP-based SAME AFSK encoder: http://www.whence.com/minimodem/

VOIP / Asterisk / Phone Interfacing

For nearly a decade I tinkered to my hearts content with various applications and packet radio. I learned an immense amount about TCP/IP. From DOS based NOS packages, which lead to ethernet networking the shack, later to Linux.

Well, I have been tinkering with Asterisk for about 6 years with the same enthusiasm. Much to my surprise, I learned that one of the original founders of our ham technology club has man of the same interests. So much love that Mike Kassner has the vanity callsign, K0PBX to reflect that.

When it comes to interacting with other people not in the hobby, messages (sometimes emergency) are relayed person (ham) to person, and the second most common way is by telephone.

As a mater of fact, telephone communication is probably the largest part of non-face to-face communication.

So like a good ham, being knowledgeable in how the telephone system works and is changing is always beneficial.

With some quick research you'll learn that SIP has become a widely adopted defacto protocol. You'll also learn that an open source Linux based application called Asterisk is a very powerful tool that many hams are playing with.

I was most pleased to notice that the second edition of Jonathan Taylor, K1RFD's VOIP book does elude to a whole new world using Asterisk and app_rpt.

In my opinion, the ham radio tie-ins are just starting.... What a perfect time to get jump on board and help develop something new.

http://nerdvittles.com/ - You'll find an immense amount of tinkering ideas here.

Southeastern Asterisk Radio Networks - Digital Amateur Projects Association (DARPA).

British Columbia Wireless Amateur Radio Network.

The BCWARN infrastructure includes and supports:

-Electronic Mail via WinLink (over 2.4ghz microwave, AX.25 packet radio and Pactor3)
-D-Star Digital voice and data
-VoiceOverIP using Asterisk private branch exchange (PBX) open source telephony switching technology
-File sharing
-Remote printing and facsimile
-Video conferencing & instant messaging

http://www.bcwarn.net

WeComm Digital Audio Repeater Linking

Wisconsin Emergency Communications (WeComm) is developing a statewide repeater system / voice network to provide base coverage throughout Wisconsin. Another imitative is to implement a modern high speed digital network. Both are for state, district and local ARES/RACES, skywarn, public service, and normal amateur radio communications.

The wide area analog repeaters are interlinked using SIP analog radio adapters. The Asterisk based conferencing software is virtually limitless. It provides the flexibility to connect individual repeaters into a statewide configuration, or to disconnect them to serve smaller areas or districts meeting specific needs.

This is a private VOIP network / reflector, unlike; IRLP and EchoLink. (Although the capability to link to such networks is still possible). Using IRLP, EchoLink or some other Wide Area Network linking services doesn't allow the control and flexibility that creating ones own off-network conferencing bridge. The aforementioned systems have a limited number of reflectors available and you are bound to their rules. Once the modern high speed digital network gets in gear the VOIP audio can actually be carried over these links instead of the internet.

SIP / digital audio repeater linking allows you to route / (think digipeat) . This is a great way to connect repeaters together that have quite a distance between them (or poor radio path).

SIP radio linking is also compatible with telephone circuits, and 21st century digital radio systems such as APCO-25, and D-Star. Such high speed multi-media interconnecting backbones can support radio linking of today as well as of the future.

Wisconsin Amateur Radio Club - Digital Backbone Project

I spotted this at Amateur Electronics Superfest. Their demo was a pair of IP phones interconnected via 2.4 GHz shotgun antennas.

They are deploying a 5.6-5.825 GHz high speed digital network in Southeast Wisconsin. They are using the Ubiquiti Bullet 5HP, a 1 watt 5 GHz capable transceiver.

Due to the 6-54 megabit bandwidth, multiple simultaneous communications can take place on a single channel. Any site can act as an intelligent digital repeater into the network and thereby expand the mesh...

Due to he 12 volt operation and the small, high gain antennas, a portable station can be quickly assembled in the field and added to the digital backbone network.

Field locations provide: VoIP telephone for command posts, real time high resolution video of conditions, e-mail and file transfers, extension of the digital network, and standard voice type QSO's.

You can view the overview handout of their digital backbone project here:

http://www.qsl.net/kb9mwr/projects/wireless/WIARC.pdf

Minneapolis Wi-Fi network aids rescuers - bridge collapse

So the real question is when are ARES/RACES groups going to realize its time to implement this using products like the Ubiquiti NS3/XR3 on 3 GHz, the lower half (2.3 GHz) of the 2.4 GHz ham band, the 5.8 GHz band, or even 900 MHz with something like the XR9.

Ubiquiti's AirOS firmware as well as popular third party firmware such as DD-WRT all now have mesh networking protocols like WDS (Wireless Distribution System) built in.

Mesh protocols are designed for self-healing networks that are able to load balance WAN (wide area network) access. They also combine some of the ideas contained in the Radio Shortest Path First protocol. If the you are not up to par on the dynamics of mesh networks, here are a few links:


http://en.wikipedia.org/wiki/Wireless_Distribution_System
http://en.wikipedia.org/wiki/Wireless_mesh_network
http://www.olsr.org/ An ad hoc wireless mesh routing daemon
You may also wish to read the article by John, K8OCL, titled "New High-Speed Multi-Media Radio Mesh Networking," printed in the Fall 2008 CQ-VHF magazine.

Mesh is very powerful stuff. And when you throw SIP / Asterisk based telephony on top of it, you have an instant voice system. Just plug in an IP phone or analog telephone adapter. Now not only can you exchange large image files, email, word-processing and other files that emergency responders and served agencies find invaluable. You can also pick up a phone or (seamlessly) bridge existing ones that are dead due to a land-line failure... etc.

From: http://www.arrl.org/tis/info/HTML/high-speed-digital/emergency-communications.html

Emergency communications offer the greatest opportunity for Radio Local Area Network (RLAN) technology to excel and for amateurs to push the envelope in the public service sector, using this technology....

Another good read is the Winter 2005 issue of CQ-VHF. John, K8OCL wrote a HSMM column on a HSMM portable setup for EmComm organizations.

D-Star <-> SIP Translation

One of the interesting things about D-Star is it's ability to route calls between radios by callsign. There are two types of calls with D-Star; directed calls and non-directed calls. Non-directed are much like we are all used to. A CQ type listening mode, where you basically hear all general CQ style traffic on the channel. In the directed call mode, you can ignore all this (think of it as call sign squelch), kind of do-not disturb mode, but either way you can be summoned by anyone making a directed call to you. The radios display the callsign of who you are talking to kind of like a caller id type of thing.

When systems are linked a gateway server tracks where various callsigns are coming from so that directed calls can route out to the appropriate radio end point as you mobile from city to city, etc.

The Session Initiation Protocol (SIP), works very similar. It has become a standard for VOIP and other text and multimedia sessions. The sip username/ secret can represent the D-Star callsign or device / radio endpoint. SIP usernames are usually represented numerically, as in extension 200, but they can also be alphanumeric. A meetme/ conference call/room can act like that big CQ style reflector that we are used to with IRLP and Echolink.

SIP devices can register dynamically with an Asterisk Sever The best example is a SIP based Wifi phone where you may travel from hotspot to hotspot, but the server keeps track of how to route calls back to you. Much like how a D-Star gateway can track and route calls to you.

In either example, D-Star or SIP, under each packets SIP data, the callsign or extension information, there is then the Real-time Transport Protocol (RTP) audio stream carrying ones voice. With D-Star this is AMBE encoded, and with SIP it's typically encoded U-law/G.711.

John, K7VE has a D-Star dream of convergence with SIP, an established real- world protocol using by VOIP networks.

To accomplish such an Asterisk/SIP to D-Star/AMBE bridge/translation, the patented D-Star codec would have to be decoded with a $20 hardware solution, such as the DV dongle developed by Moe Wheatley, AE4JY. It can then be transcoded to more common g.711 a-law/u-law codec.

He points out that once a channel driver is properly created the whole power of Asterisk can be brought to play as a D-Star radio can then be used like any other digital IP phone endpoint; conference bridges, interactive voice response, call out, autopatch, voice recognition, etc. Basically instead of an IP phone as a digital endpoint, we could use a D-Star radio.

Using the AMBE-2020 chips in a PCI card or USB dongle will allow the conversion of the DSTAR Digital Voice to/from alaw or ulaw 8k digital voice and the chip decodes/encodes DTMF ... this combined with the datastream (containing callsigns) should enable making DSTAR radios extensions on a Asterisk PBX (or even assign a DID to them) -- total 2-way ROIP/VOIP integration that can route to/through the PSTN. (Example: dial 1-800-4KB9MWR and get a call you can pick up on your DSTAR radio and vice-versa.) Pretty powerful for EMCOMM and personal use. All enabled by open source

http://www.mail-archive.com/dstar_digital@yahoogroups.com/msg02975.html

-Update May 1st, 2009-
Scott, KI4LKF announced today that he has written an asterisk channel driver (chan dstar) that can connect to D-STAR repeaters and reflectors.

Keep your eyes peeled as this is further developed. http://asteriskradio.net/wp/2009/04/29/digital-integration-to-pstn/

And here are some thoughts on using a SIP phone/client on an Iphone or Android phone to connect to D-Star.

Here is a bit more of John's thinking. A cut and paste from an inquiry to the rtpDir yahoo group:

Subject: Re: Question on Dstar-Asterisk Integration
Date: Thursday, July 23, 2009 3:56 PM



--- In rtpDir@yahoogroups.com, "Tom Power" wrote:

Question for the Group:

Does Scott, KI4LKF's Dstar to Asterisk integration allow the passing of the Dstar integrated callsign in the Dstar Stream to Inbound Caller ID on Asterisk/SIP?

Wondering the level of integration with Asterisk.

Thinking of doing some experimenting....
Was thinking on a PBX integration point. Basically two ID-1's with one of the ID-1's having an asterisk server on one end and using it as a Autopatch, announcement system, SMS Gateway etc.

Thanks,
Tom

As Scott said, that is not what his code does, but it is something several folks have a strong interest in seeing implemented.

Now that open source (Scott's and G4ULF) gateway implementations are coming about, the integration should be a lot more approachable.

Here is what I would like to see.

A driver on Asterisk that has the following functionality:

* It creates a channel to D-STAR where an extension or DID could be mapped to a D-STAR callsign. This channel would do a few things:

Route the stream through an AMBE device (such as the DV Dongle) and bi-directionally transcode between the source CODEC (GSM, G.711, etc.) and AMBE.

It would initiate a "ring" to the D-STAR network through a gateway, setting the UR field to the destination callsign and the MY field to a callsign assigned to the PBX (e.g. KX1XXX T - T for telephony interconnect). The "caller ID" would be placed in the Icom defined 20 character short message field, so that it would show up on the display of D-STAR radios on the radio side of things. It could also send an "audio" indicator of a ring in the stream.

When the radio receives the "ring", the radio user sends back a DTMF command such as "*" to answer, "#" to hangup, "1" send to voicemail, "2" send to my landline, etc.

If no command is received within a certain amount of time, the call is sent to Voicemail.

The conversation takes place (with predefined timout).

If there is also text from the same session (e.g. Jabber/SIP) it is put into the data stream on D-STAR, so it will be available on the data port on the radio. (Bi-Directional)

At the end of the conversation, the radio user sends a DTMF hangup command (e.g. "#") - since this is connectionless, if the "telephone" user hangs up, the DTMF is unnecessary.

Conversely the radio user, sets up a call by setting up his radio:

UR: KX1XXX T
RPT1: KX1XXX A (the repeater)
RPT2: KX1XXX G (the gateway -- I think this should be removed in the future, but its how it works now)
MY: K7VE

The radio user sends a DTMF sequence to start the patch, such as "*123", then dial the number/extension you want to call and proceed like a "regular" telephone call. Then hangup using a DTMF command.

BTW, all of this should work through the D-STAR network. So you could actually go to a remote "registered" Telephony Interconnect. E.g:

UR: KX1XXX T (Boston Interconnect)
MY: K7VE (I'm in Seattle)
RPT1: K7LWH C (2 meter Bellevue WA Repeater)
RPT2: K7LWH G (Bellevue WA Gateway)

Pretty cool?

Once this is all working it is simple to use the Asterisk PBX for voicemail, interactive voice response, etc.

I am gathering the components to build a "desktop repeater" for testing, at which point I hope to work on this.

-- John, K7VE

http://www.mail-archive.com/dstar_digital@yahoogroups.com/msg02975.html

{Edit 1/2011}
Karl, N2VQM posts some good thoughts on the whole matter:
http://groups.yahoo.com/group/gmsk_dv_node/message/6210

DV Dongle

If you have been following my blog you probably say my piece on interoperability strides and my other piece on this second roll out of D-Star called NXDN.

This interoperability thing as you can imagine is pretty important. The key problem stems from a lack of standards but it really has to do more so with the fact that technology is developing so quickly that it's difficult to get everyone on the same page. Technology standards seem to be set by who can develop something first and obtain the largest market following. Just look back at VHS vs Beta, Blueray vs HD-DVD, etc.

As pointed out a common interconnection that both P25, and D-Star developers are choosing is the standardized Asterisk based SIP protocol using RTP audio streams.

A key component in all these digital voice systems is the vocoder. This is the device that converts your spoken voice to a synthesized or digitally compressed speech format.

The public safety APCO/ P25 format uses Improved Multi-Band Excitation (IMBE). This is proprietary vocoder developed by Digital Voice Systems, Inc. (DVSI). It is the predecessor of their Advanced Multi-Band Excitation (AMBE). It costs $150K to get the rights to play with that mode plus $5 a seat. There is no off the shelf IC to do it.

D-Star uses Advanced Multi-Band Excitation (AMBE) from DVSI. Same $150K if you want the software source but they do offer a single chip solution for $20 single quantity and are happy to sell to hams.

The DV dongle is an important development. It was started by Moe, AE4JY and Robin, AA4RC. It contains the ability to process AMBE full duplex. Presently software applications exist to use this to communicate from a computer to a D-Star gateway. Further development are expected so that it can be interfaced to a radio's packet radio port that has the necessary discriminator connections. This may be a huge milestone. The ability to retrofit an existing repeater could be possible with this. Not only that, but you may be able to retrofit it in such a way that it can be usable in analog and for D-Star. 

With the DV Dongle you will be able to use DPLUS* or the future OpenDSTAR gateway software to connect to the gateway computer behind the ID-RP2C (repeater controller). Then you will use the DV Dongle to extract the audio streams and can transcode them to standard G.711a/u, GSM, G.729, etc. and then use the Asterisk PBX power for telephone, voicemail, DID-Callsign-DID, repeater, etc. interconnection.

(* DPLUS is a gateway addon daemon that provides a number of functions see: http://opendstar.org/tools/readme.txt)

When you take a digital radio platform like D-Star, this is where integrating Asterisk could be very powerful. Since the call sign is part of every packet, this could be assigned to a direct inward dialing number (DID) or extension.

At the present time the dongle connects over the internet to an Icom gateway controller at a repeater site. So for now is a non-RF application.

What would be even more ideal since D-Star radios don't have an analog packet port is if the various D-Star radios had a digital interface port/jack to support just such a dongle / device to transcode to a SIP and codec standard. Unfortunately at this time there are no known interfaces to the Icom D-STAR radios that allow access to the on-air data stream.

I'd love it if I could buy a D-Star radio, that has an digital interface of sorts. Something along the lines of D-Star non-proprietary interface. Perhaps a mini-USB interface that perhaps transcodes to a more open standard codec like G.711 using SIP and RTP protocols (standardized protocols) so one can connect the radios together into wide area networks.

Then one would be able to have a D-Star radio in my shack also interconnected to various Asterisk powered applications in their house. Where if you weren't around to take a directed D-Star call, it could be configured as a DID to a system and use a ring group / follow me list to let the radio caller ring a home phone or leave a voice mail.

The open source project25 interface is a good idea. Unfortunately the problem they're facing is that most of the manufacturers don't bother following the ISSI spec, nor does the ISSI spec call out hardware interface details. So basically the "plugs" on the back of say, a Quantar... don't match the "plugs" on the back of a MASTR III.

http://en.wikipedia.org/wiki/P25_ISSI

The Project 25 Inter RF Subsystem Interface (P25 ISSI) is a non-proprietary interface that enables RF subsystems (RFSSs) built by different manufacturers to be connected together into wide area networks.

It would be great to see an a D-Star radio that supports something like this on the market possibly before the P25 interface idea ever makes it to the market.  

ARRL: It Seems to Us: Interoperability October 1, 2007 We need to encourage D-Star manufactures to come up with a similar style non-proprietary interface for D-Star.  

Digital Voice Interoperability strides

Asterisk provides an Interoberabity platform

What is Asterisk?
Asterisk is an Open Source PBX & Telephony Platform. It’s often labeled as the future of telephony. It has been around since about 1999 and the platform is open source - Linux operating system based. It can support a variety of signaling protocols, but by far SIP, the session intitation protocol, for VOIP and other text and multimedia sessions, has become a standard.

For those scratching their head a bit... PBX stands for private branch exchange. It is a machine that handles many businesses telephones calls for you. Its main functions are to transfer calls to different individual phones; play music when somebody is put on hold; to play automated voice responses when a call is received; to provide an options menu for the caller etc.

Asterisk allows one to build their own phone systems. It adds features, functionality and reduces deployment costs in ways which; at first are a little difficult to understand.

How does this relate to amateur radio?

Very simple, the future of two way radio is digital. As of writing, TV are required to be full digital and shut down their analog transmitters in Feb. 2009. The only spectrum broadcasters are required to vacate are channels 64 thru 69 that will become the new "700 MHZ band" that is being auctioned off by the FCC. The vacated areas of this spectrum will be utilized for: Public Wireless deployment (Cellular/PCS); A wide-band private data network that will be shared between public safety and paying customers; and new spectrum for public safety that will butt right up to the re-located NPSPAC National Public Safety Planning Advisory Committee band being moved to 806-809/851-853 by Sprint/Nextel.

Public safety also has guidelines to migrate to APCO-25 digital. The future of two way radio is digital, and we must also advance in this direction. The digital premise is that it generally allows more use in a more efficient/flexible use of band space.

Most present day government communication centers that use analog systems happen to have a VOIP based dispatch console. This analog to VOIP patching is something that we are presently also embracing in ham radio with IRLP, EchoLink, Yeasu WIRES II, and the like.

Your seeing the digital migration in the commercial world as I pointed out; And the only analog part left of traditional telephone is the “last mile” drop to your home. Time Warner and now AT&T are providing digital phone service to close that up too.

As of writing there aren't any 100% directed approaches to tie this to the hobby that I can point you to. There are a number of open ended ideas from a variety of different people. What I'm saying is there is no one entity steering the ship, so to speak. This ideas are still in development. Which makes it precisely the time to jump aboard and get our hands in it and see what we can do with it. So in light of that I suggest a google search for more info...
Digital Voice Interoperability Software Strides
Perhaps some of you remember the ARRL "It Seems to Us: Interoperability" statement from October last year regarding emerging digital voice.

Well the good news is here are some of the strides I've run across:

-OpenP25 Project http://openp25.org/index.php/category/project-status/

"We've determined that the open source Asterisk PBX appears to be a good framework on which to build a P25 ISSI (Inter RF Subsystem Interface) switch. Asterisk has a mature SIP stack and already has the ability to transparently pass RTP frames between SIP channels. The National Institute of Standards and Technology (NIST) has made an open source program for ISSI testing freely available to the P25 community. A full-featured open source P25 ISSI switch is clearly achievable."

The Project 25 Inter RF Subsystem Interface (P25 ISSI) is a non-proprietary interface that enables RF subsystems (RFSSs) built by different manufacturers to be connected together into wide area networks. Apparently consideration is being given to enhancing Asterisk app_rpt to support a such a low-level P25 radio interface.
The open source project25 interface is a good idea. Unfortunately the problem they're facing is that most of the manufacturers don't bother following the ISSI spec, nor does the ISSI spec call out hardware interface details. So basically the "plugs" on the back of say, a Quantar... don't match the "plugs" on the back of a Mastr III.

-Open D-Star Project http://opendstar.org/

The OpenDSTAR group has released several software tools which build on existing commercially available repeaters and Internet gateways to extend functionality.

And this interoperability work in progress attempt has been around longer than the P25 one. They too are working a SIP stack into their DPLUS gateway add-on daemon.
A project is lead by Scott, KI4LKF expand the Asterisk capabilities with his RtpDir bridge software (Real TimeProtocol Director) software package for VoIP/RF Gateways. http://tech.groups.yahoo.com/group/rtpDir/
It can bridges transmit and receive from/to Asterisk/app_rpt, IRLP, Echolink, Speak-Freely and he is working on D-Star. There is support for all link interfaces, sound mode or ASCII mode, VA3TO, WB2REM, ULI, Rigblasters, MFJ, SignalLink, etc.
-RadioGrid RG-4 Radio Gateway. http://www.radiogrid.com/
The guys from NHRC who make repeater controllers have come up with a commercially targeted (and priced) controller called the RadioGrid RG-4 Radio Gateway. It's built around Asterisk open source PBX technology. It's specifically for VOIP linking designed with interoperability applications in mind. It uses a 500 MHz Blackfin processor, 64 MB RAM, 4 MB Flash memory, 1 GB SD Card

A key component in all these digital voice systems is the vocoder. P25 uses the IMBE vocoder from Digital Voice System Inc. (DVSI). It costs $150K to get the rights to play with that mode plus $5 a seat. There is no off the shelf IC to do it. Fortunately D-Star uses AMBE from DVSI they do offer a single chip solution for $20 single qty and are happy to sell to hams.

- The DV Dongle http://dvdongle.com/
The DV dongle is an important development. It was started by Moe Wheatley, AE4JY and Robin Cutshaw, AA4RC. It contains the hardware ability to process AMBE full duplex. Presently software applications exist to use this to communicate from a computer to a D-Star gateway. Further development are in the works so that it can be interfaced to a radio's packet radio port that has the necessary discriminator connections. This may be a huge milestone. The ability to retrofit an existing repeater could be possible with this. Not only that, but you may be able to retrofit it in such a way that it can be usable in analog and for D-Star.

It would be great to see an a D-Star radio that supports something like this on the market possibly before the P25 interface idea ever makes it to the market. We should to encourage D-Star manufactures to come up with a similar style non-proprietary interface for D-Star. As unfortunately at this time there are no known interfaces to the Icom D-STAR radios that allow access to the on-air data stream.

It would be nice to see some agreement and standards in place amongst the various guys working on software solutions and the manufactures, to help make things more stream-lined. Either way people are working on interoperability solutions, toward convergence with open standard codecs like G.711 using real word protocols like SIP and RTP protocols so one can connect the radios together into wide area networks.

Then one would be able to have a D-Star radio in their shack also interconnected to various Asterisk powered applications in their house or abroad. Where if you weren't around to take a directed D-Star call, it could be configured as a DID to a system and use a ring group / follow me list to let the radio caller ring a home phone or leave a voice mail. Calls could be route to/through the Public switched telephone network and so on. A very powerful ability for EMCOMM and personal use.

What is Asterisk?

Asterisk is an Open Source PBX & Telephony Platform. It’s often labeled as the future of telephony.

PBX stands for private branch exchange. It is a machine that handles many businesses telephones calls for you. Its main functions are to transfer calls to different individual phones; play music when somebody is put on hold; to play automated voice responses when a call is received; to provide an options menu for the caller etc.

Asterisk allows one to build their own phone systems. It adds features, functionality and reduces deployment costs in ways which; at first are a little difficult to understand.

How does this relate to amateur radio?

Very simple, the future of two way radio is digital. As of writing, TV are required to be full digital and shut down their analog transmitters in Feb. 2009. The only spectrum broadcasters are required to vacate are channels 64 thru 69 that will become the new "700 MHZ band" that is being auctioned off by the FCC. The vacated areas of this spectrum will be utilized for: Public Wireless deployment (Cellular/PCS); A wide-band private data network that will be shared between public safety and paying customers; and new spectrum for public safety that will butt right up to the re-located NPSPAC National Public Safety Planning Advisory Committee band being moved to 806-809/851-853 by Sprint/NEXTEL.

Public safety also has guidelines to migrate to APCO-25 digital. The future of two way radio is digital, and we must also advance in this direction. The digital premise is that it generally allows more use in a more efficient/flexible use of band space.

Most present day government communication centers that use analog systems happen to have a VOIP based dispatch console. This analog to VOIP patching is something that we are presently also embracing in ham radio with IRLP, EchoLink, Yeasu WIRES II, and the like.

A different hardware board for each of these proprietary VOIP systems that you want to support is required. You also need a need a multi-port repeater controller, to support each hardware boards analog breakout. This seems redundant to me, and is something that slows the advancement. IRLP seems to be the system of choice because it runs on the Linux operating system. This is because Linux is much more stable that Windows, and is an open source development.

Your seeing the migration in the commercial world as I pointed out; hello digital TV. And the only analog part left of traditional telephone is the “last mile” drop to your home. Time Warner and now AT&T are providing digital phone service to close that up too.

I really feel there "Could be" something big with Asterisk Telephony and perhaps D-Star. The marriage seems natural. I even think it can be integrated with existing VOIP systems like D-Star and EchoLink.

I feel anything is only a "could" type of thing, only because of how the concepts are presented to the amateur audience. This hobby is supposed to be about advancing technology...

As of writing there aren't any directed approaches to tie this to the hobby that I can point you to. There are a number of open ended ideas from a variety of different people. What I'm saying is there is no one entity steering the ship, so to speak. This ideas are still in development. Which makes it precisely the time to jump aboard and get our hands in it and see what we can do with it. So in light of that I suggest a google search for more info... Once you get interested you're likely to bump into myself or other hams on various message boards. And you will likely also have run across a few ideas on how to integrate it to the hobby.

If your interested in giving Asterisk a test drive I found this video overview a good starting point for myself. AsteriskNOW, or PBX in A Flash are both good starting places. They are a Linux install with Asterisk and a Asterisk GUI rolled into a bootable ISO CD install.