[asterisk-dev] Development of asterisk 1.4.23 Can we please get some development?

[Tim S]

I'd think at this point it would be less effort in total to move to a
different method for repeater operation.

What about a RaspberryPi (or equivalent) running Python/Linphone?  Using a
cheap USB 1.1 compliant audio dongle, plus the onboard GPIOs, you would
have enough resources for two radios in a single $40 board.  This gets you
away from needing a separate radio logic board as you can hit and read
GPIOs directly, and probably lowers your repeater power demand with no more
local "PC", additionally you can run the bridge on a battery much easier
than a desktop PC.

How I've been playing with Radio-over-IP (RoIP) is a setup as described
above, where the RaspberryPi dials into a conference bridge as the host,
and only one talker is allowed in a conference - giving a busy tone to
other talkers if they try to "grab" the conference mic - similar to what a
Nextel PTT does while it's actively receiving on the channel.  The
RaspberryPi does not need an active conference to work as a repeater, it
can store and forward (re-TX) the audio without ever involving VoIP all
inside Python.  I have no HAM license, so my experiments are limited to my
Faraday cage (some day I'll take the tests, and go outside).  In my
opinion, this method made RoIP most effective, as all of the features big
distributed radio system want can be applied this way, even multi site
distributed radio and shared radio single transmitter systems (think
commercial/military and emergency dispatch respectively).

Moving the actual radio control into a RaspberryPi and Python/Linphone,
makes it easier to do other things like talk to radios over a serial
programming link (if available), the libLinphone library sends all of the
out-of-band DTMF signals to the Python core - where "the sky is the limit"
on what you program to act on those tones.  Most of the repeater "magic"
happens in Python.  You could even use a re-encode as a digital mode for
multi-hop repeating, where the first repeater gets a request to hop to a
very distant repeater, and the first repeater can trunk the transmission
over TCP/IP-SIP (private radio networks), or push it as digital audio
(Codec2/FDMDV) with decent error correction to the distant repeater, which
then finally re-broadcasts it as analog audio.

I did plan on checking the little change this required into the current
Asterisk dev-tree once I made the code pretty.  Only a single and slight
change was needed to the existing conference app (I just added one
configuration setting identifying it as a RoIP bridge) - which enables the
special RoIP-conference mute logic.  My intention was to minimize the
special code requirements within Asterisk, and generalize radio control
more into a phone-friendly operation and call flow when radio traffic
needed to be bridged.

If say a radio manufacturer felt compelled, they could write a binary blob
to drop into Python to control their radios over a serial/digital link and
keep their commands proprietary - or they could simply drop a better $2 ARM
processor with a network PHY in their radio with the FOSS libLinphone and
python to natively run the radio core and just give people an Ethernet port
straight into the radio's DSP (at this point, I have no idea why this isn't
the standard way of doing things anyway when a radio costs over $100).

Some other ideas I had for that was stacking two RaspberryPi's to acts as
high availability RoIP bridges - the two R-Pi's being connected to two
radios, and talking to each other over both TCP and a local serial bus.  If
either stopped working the other could take over the functions of the dead
one, and together they could trouble-shoot down to a dead radio.

Other things you could do by putting more logic in the Python code outside
of the radio but before the libLinphone library:  by hooking in digital SWR
meter on the antenna feed out of the radio you could do some basic
troubleshooting of the antenna, put in a volt meter and you can watch
remote repeater site batteries, check fuse conditions (blown/good), measure
amplifier current, solar panel collection power.  Just an idea as food for
your thought...

-Tim

On Thu, Jul 14, 2016 at 7:57 PM, <asterisk-dev-request at lists.digium.com>
wrote:

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> Today's Topics:
>
>    1. Re: Development of asterisk 1.4.23 Can we please get some
>       development? (Jonathan Rose)
>    2. Re: Development of asterisk 1.4.23 Can we please get some
>       development? (Loren Tedford)
>    3. Re: Development of asterisk 1.4.23 Can we please get some
>       development? (Kirill Marchuk)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 14 Jul 2016 17:12:58 -0500
> From: Jonathan Rose <jonathan.rose at motorolasolutions.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] Development of asterisk 1.4.23 Can we
>         please get some development?
> Message-ID:
>         <CABkuVg+ZV_XSgDk_QUWRr80YJEN=
> ZGGfK1VNS-aU+zzNBqLmvg at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Thu, Jul 14, 2016 at 3:27 PM, Loren Tedford <lorentedford at gmail.com>
> wrote:
>
> > First off i want to say thanks for the fast replies and want to say no I
> > don't plan on putting out a job offer.. But was just interested in some
> > help in general of putting in at least the basics we require into
> > asterisk.. You guys have moved so far on and forgotten about us back here
> > in the stone age.. Its ok we will keep working on it and hopefully we get
> > some where.. It was only just a thought that maybe some of you all would
> be
> > interested in at least helping release a security patch for 1.4.23 on
> some
> > of the known issues of back in that time.. Thanks again..
> >
>
>
> Having been in the midst of a similarly complicated upgrade procedure for a
> custom branch of Asterisk for the past few months myself, I get the
> frustration... but it's not that you were forgotten. All of the releases
> were made available to you and you had the opportunity to address breaking
> changes incrementally rather than letting it build into a monster. You all
> are the ones who forgot about the software you were using and let it build
> up technical debt for ten years.
> --
>
> *Jonathan R. Rose*Senior Systems Engineer
>
> Emergency CallWorks
> Motorola Solutions
>
> email: jonathan.rose at motorolasolutions.com
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> ------------------------------
>
> Message: 2
> Date: Thu, 14 Jul 2016 17:25:43 -0500
> From: Loren Tedford <lorentedford at gmail.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>,
>         jonathan.rose at motorolasolutions.com
> Subject: Re: [asterisk-dev] Development of asterisk 1.4.23 Can we
>         please get some development?
> Message-ID:
>         <CAK=eTyg6WtuFYHCn=Jt+fOKQr0=
> AVv1i3OMaBjxi57EhJwGg-Q at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> I have only been using the project since 2013 I had no idea at the time
> what version of asterisk was being used to install the modules needed..
>
> Forgotten or not this is where we stand with it now..
>
> Even if a developer donated 20 minutes on it a week that adds up to roughly
> 17 hrs of possible improvement from what we got..
>
> I have experimented with trying to cobble things together on older versions
> of asterisk what i generally find is it literally barfs all over the
> place.. It really seems like they have changed some thing in the way
> asterisk handles the transmit and receive sides of everything.. To my
> knowledge I do not see any developers with  in the app_rpt actually posting
> any work or upgrades to the project.. I figured since asterisk was asterisk
> at one point in time developers might at least be interested in helping out
> or donating some time..
>
> Their is alot of cool things that could be done with it if it was
> integrated with asterisk phone system today even in the commercial world..
>
> I wonder what exact core changes occurred to create the problem we have
> today and is it reversible or do we need to completely redesign asterisk..
>
>
> Loren Tedford (KC9ZHV)
> Phone:
> Fax:
> Email: lorentedford at gmail.com
> Email: KC9ZHV at KC9ZHV.com
> http://www.lorentedford.com
> http://www.kc9zhv.com
> http://forum.kc9zhv.com
> http://hub.kc9zhv.com
> http://Ltcraft.net <http://ltcraft.net/>
> http://voipham.com
>
> On Thu, Jul 14, 2016 at 5:12 PM, Jonathan Rose <
> jonathan.rose at motorolasolutions.com> wrote:
>
> > On Thu, Jul 14, 2016 at 3:27 PM, Loren Tedford <lorentedford at gmail.com>
> > wrote:
> >
> >> First off i want to say thanks for the fast replies and want to say no I
> >> don't plan on putting out a job offer.. But was just interested in some
> >> help in general of putting in at least the basics we require into
> >> asterisk.. You guys have moved so far on and forgotten about us back
> here
> >> in the stone age.. Its ok we will keep working on it and hopefully we
> get
> >> some where.. It was only just a thought that maybe some of you all
> would be
> >> interested in at least helping release a security patch for 1.4.23 on
> some
> >> of the known issues of back in that time.. Thanks again..
> >>
> >
> >
> > Having been in the midst of a similarly complicated upgrade procedure for
> > a custom branch of Asterisk for the past few months myself, I get the
> > frustration... but it's not that you were forgotten. All of the releases
> > were made available to you and you had the opportunity to address
> breaking
> > changes incrementally rather than letting it build into a monster. You
> all
> > are the ones who forgot about the software you were using and let it
> build
> > up technical debt for ten years.
> > --
> >
> > *Jonathan R. Rose*Senior Systems Engineer
> >
> > Emergency CallWorks
> > Motorola Solutions
> >
> > email: jonathan.rose at motorolasolutions.com
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
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> ------------------------------
>
> Message: 3
> Date: Fri, 15 Jul 2016 09:57:42 +0700
> From: Kirill Marchuk <62mkv at mail.ru>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Development of asterisk 1.4.23 Can we
>         please get some development?
> Message-ID: <18eed78a-2fa8-1f32-c94e-b471ed1cd6b8 at mail.ru>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> Can the whole problem be reformulated as "app_rpt does not work with
> Asterisk 13+" or is there anything more than that ?
>
> If the former, did you try with any of the 1.8.x version ?
>
> Kirill
>
> 15.07.2016 5:25, Loren Tedford ?????:
> > I have only been using the project since 2013 I had no idea at the
> > time what version of asterisk was being used to install the modules
> > needed..
> >
> > Forgotten or not this is where we stand with it now..
> >
> > Even if a developer donated 20 minutes on it a week that adds up to
> > roughly 17 hrs of possible improvement from what we got..
> >
> > I have experimented with trying to cobble things together on older
> > versions of asterisk what i generally find is it literally barfs all
> > over the place.. It really seems like they have changed some thing in
> > the way asterisk handles the transmit and receive sides of
> > everything.. To my knowledge I do not see any developers with  in the
> > app_rpt actually posting any work or upgrades to the project.. I
> > figured since asterisk was asterisk at one point in time developers
> > might at least be interested in helping out or donating some time..
> >
> > Their is alot of cool things that could be done with it if it was
> > integrated with asterisk phone system today even in the commercial
> world..
> >
> > I wonder what exact core changes occurred to create the problem we
> > have today and is it reversible or do we need to completely redesign
> > asterisk..
> >
> >
> > Loren Tedford (KC9ZHV)
> > Phone:
> > Fax:
> > Email: lorentedford at gmail.com <mailto:lorentedford at gmail.com>
> > Email: KC9ZHV at KC9ZHV.com <mailto:KC9ZHV at KC9ZHV.com>
> > http://www.lorentedford.com <http://www.lorentedford.com/>
> > http://www.kc9zhv.com <http://www.kc9zhv.com/>
> > http://forum.kc9zhv.com <http://forum.kc9zhv.com/>
> > http://hub.kc9zhv.com <http://hub.kc9zhv.com/>
> > http://Ltcraft.net<http://ltcraft.net/>
> > http://voipham.com
> >
> > On Thu, Jul 14, 2016 at 5:12 PM, Jonathan Rose
> > <jonathan.rose at motorolasolutions.com
> > <mailto:jonathan.rose at motorolasolutions.com>> wrote:
> >
> >     On Thu, Jul 14, 2016 at 3:27 PM, Loren Tedford
> >     <lorentedford at gmail.com <mailto:lorentedford at gmail.com>> wrote:
> >
> >         First off i want to say thanks for the fast replies and want
> >         to say no I don't plan on putting out a job offer.. But was
> >         just interested in some help in general of putting in at least
> >         the basics we require into asterisk.. You guys have moved so
> >         far on and forgotten about us back here in the stone age.. Its
> >         ok we will keep working on it and hopefully we get some
> >         where.. It was only just a thought that maybe some of you all
> >         would be interested in at least helping release a security
> >         patch for 1.4.23 on some of the known issues of back in that
> >         time.. Thanks again..
> >
> >
> >
> >     Having been in the midst of a similarly complicated upgrade
> >     procedure for a custom branch of Asterisk for the past few months
> >     myself, I get the frustration... but it's not that you were
> >     forgotten. All of the releases were made available to you and you
> >     had the opportunity to address breaking changes incrementally
> >     rather than letting it build into a monster. You all are the ones
> >     who forgot about the software you were using and let it build up
> >     technical debt for ten years.
> >     --
> >     *Jonathan R. Rose
> >     *Senior Systems Engineer
> >
> >     Emergency CallWorks
> >     Motorola Solutions
> >
> >     email: jonathan.rose at motorolasolutions.com
> >     <mailto:jonathan.rose at motorolasolutions.com>
> >
> >     --
> >     _____________________________________________________________________
> >     -- Bandwidth and Colocation Provided by http://www.api-digital.com
> --
> >
> >     asterisk-dev mailing list
> >     To UNSUBSCRIBE or update options visit:
> >     http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> >
> >
>
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