[ale] OT Asterisk user contacts

Fri Nov 16 11:15:42 EST 2018

I started at the same point where you're at now with my home setup and
eventually went full SIP, although I do not use fax at home. This is with
FreePBX running on an old notebook. I also started out with those little
Linksys SPA units. They can be a little flakey but can be coerced. I never
got them to handle fax well (at least not SIP-to-fax).

At work, I have had a degree of success using Audiocodes devices when it
comes to fax. I use these in transparent mode for converting SIP calls to
analog, and the fax is then delivered to a copier with a built in fax.
Audiocodes can be pricey but if you trawl ebay you can probably find
something reasonably priced. The Audiocodes registers as a standard SIP
extension against FreePBX. Be warned though that the different models of
Audiocodes (and versions of their firmware) differ in subtle ways and a
single setting, just slightly off, can be the difference between it working
and not working. Be ready for some frustration getting the hang of it.
Documents like this have been very helpful to me
https://nachbar.name/2011/06/29/audiocodes-mp-112-fax-with-asteris/

If you plan to eventually use SIP trunks, you will need a better quality
device, like and Audiocodes for fax. Fax over SIP seldom "just works".

I use the MP202B FXS for SIP to Fax: e.g.
https://www.ebay.com/itm/MP202B-2FXS-AUDIOCODES-Analog-Media-Gateway-VoIP-Adapter-MP202B-2S-SIP-NEW/283156735055?epid=1637991410&hash=item41ed74e44f:g:25wAAOSwQS1bmW4N:rk:3:pf:0

I guess I'm coming at it more the SIP trunk angle, but in my experience,
all this experimentation you're currently doing is the precursor to taking
your home PBX project to full SIP.

Too bad GoogleVoice has recently killed off the ability to work on FreePBX
(well it does still kind of work, but only GoogleVoice to GoogleVoice).
That was a nice way to have a free home setup. But my current system with
pay-as-you-go SIP has saved me a lot over the years compared to Comcast's
$30 a month deal. As for the time I invested in getting there ... LOL ... I
guess I'll never recover that, but it's been fun.

If you have a old PC that you can sacrifice for Asterisk, I have a brand
new in-box Sangoma A200 Analog FXO/FXS card (note: PCI bus - not PCI-e)
that you can have for free. This *could* make your life a bit easier in
some respects. But it could also open a new door to hell  ... :-)

If you want it you'd need to come pick it up at my office in Johns Creek.

cheers
ed

_________________________________________

*Edward O. Holcroft*
IT Operations Manager

*Madsen, Kneppers & Associates, Inc.*
Construction Consultants & Engineers
11695 Johns Creek Parkway, Suite 250
Johns Creek, GA 30097

*O*  770.446.9606  |  *F*  770.446.9612  |  *C*  770.630.0949  |
eholcroft at mkainc.com

www.mkainc.com

On Thu, Nov 15, 2018 at 7:14 PM Alex Carver via Ale <ale at ale.org> wrote:

> On 2018-11-15 13:47, Joey Kelly via Ale wrote:
> > On Wednesday, November 14, 2018 10:53:58 PM EST Alex Carver via Ale
> wrote:
> >> On 2018-11-14 18:13, Edward O. Holcroft wrote:
> >>> Why not just ask your question here? Or is that not permitted?
> >>>
> >>> Asterisk is hardly that a long stretch OT for a LUG. I'd bet most of
> the
> >>> ex-AAUG folks are on this list.
> >>
> >> I think it's just a bit overly Asterisk specific and likely to drag on a
> >> bit so it would overextend the welcome of an OT thread
> >
> > You mean like this one? ;-)
> >
> > Seriously, start a thread and ask your questions, maybe I/we can help
> you out.
> >
>
> Hah, this one hasn't gone on too long with all sorts of sordid, detailed
> information :)
>
> I've gotten some feedback from Scott and Derek but I'd be happy to send
> along what I wrote to them.  You can skip way down to the
> TL;DR section if you like as well.
>
> I've been playing around with Asterisk at home as a hobby for a little
> while.  Not too long ago my father-in-law had a small pile of SPA942's
> laying around and gave them to me which prompted me to try and install a
> nice VoIP system at home and it went "downhill" from there.  It's been a
> lot of fun tinkering with Asterisk (no GUI, just CLI with raw Asterisk)
> and doing all sorts of weird things (my laundry machines send text
> messages to the 942's when the washer or dryer is done with a load :) ).
>
> Up until this point, though, the system can't dial out anywhere but I'm
> not ready to pony up for paid SIP trunking with a SIP provider yet.  So
> I figured I would try to use my PTSN lines that I already pay for as
> part of my ISP bundle as the trunk.  I started looking at network
> attached gateway devices (I run Asterisk in a VM currently and will
> eventually put it on a SBC like a Raspberry Pi).  The one I kept running
> across a lot was the Linksys SPA3102 which had two FXO and two FXS
> ports.  I didn't realize how old that device was and that it really was
> no longer supported or even really sold except as old "new stock" or
> used.  The main thing was trying to avoid breaking the bank since this
> is a hobby and for personal use but if it's unavoidable then I'd just
> have to save pennies.
>
> I wasn't really sure if support was really required but I figured better
> to be safe with somewhat newer hardware that still gets manufacturer
> support but I don't know what to look for or what currently is suitable
> both as a manufacturer (I know I hear a lot about Cisco devices giving
> Asterisk people headaches) and as a piece of hardware.
>
> The basic things I'm looking for that I really want to be supported in a
> gateway device would be:
>
> * Supporting the FXO as a SIP trunk (this seems to be a given)
>
> * Passing PTSN caller ID data into the Asterisk system so I can send
> that along to all phones
>
> * Indication of line busy status (for BLF purposes)
>
> * Ensuring I/Asterisk know which phone line is calling inbound for
> routing purposes (this is because in the future I may end up with a
> second PTSN line and would need to route calls to the correct station)
>
> * At least one FXS port so I can keep my cordless phone going (though if
> really needed I'd consider an independent FXS adapter if there wasn't a
> built-in FXS port).
>
> * Here's the fun one: support hook-flash so I can handle call waiting on
> inbound calls from the SIP phones (somehow convincing Asterisk to send
> the flash message to the gateway and have the gateway do it).  This
> seems hard to do just from Google searches and may have to end up on a
> nice to have list with possible workarounds.
>
>
> Nice to haves were:
>
> * FXO to FXS fail-over for emergencies though I do plan to have at least
> one hard phone tied into the PTSN line anyway, it'll just be out of the
> way as the cordless and desk SIP phones would be the primary units.
>
> * Supporting two FXOs on a single device (I could deal with two
> independent devices but it would be nice not to eat too much shelf space)
>
> * Support for handling a fax modem (my current ISP device handles faxes
> ok, I think it does pass through though it claims it also handles T.38).
>  For this I already have a standard (old) Supra fax modem hanging off
> another Raspberry Pi running hylafax.  I would consider using IAXmodem
> with hylafax to stream the data into Asterisk and then pass through the
> gateway but initially the fax modem is going to be hanging out on an FXS
> port along with my analog phones.
>
> * Support for sending Caller ID data out the FXS port so the cordless
> phones still receive CID info.
>
> I was viewing some reviews on YouTube for other things and ran across a
> Sangoma Vega gateway (four FXS, four FXO) that seemed to be an option
> but rather expensive at around $300 or so and probably does everything
> (haven't found a manual yet).  I also heard about some stuff from
> Grandstream but then some of the reviewers have made complaints about
> Grandstream before so it's unclear if that's a good option.
>
> The TL;DR:
> I want to be able to make outgoing calls from my VoIP system and use the
> PTSN as my primary trunk so I need an affordable gateway device that can
> at least handle Caller ID data to feed into the Asterisk system and also
> run my analog lines as SIP devices (with CID as well) and can hopefully
> support Call Waiting with a hook-flash message from Asterisk.  I spotted
> a Sangoma Vega that might work but is expensive (~$300) and was told
> that Grandstream has some devices as well but am unsure whether
> Grandstream is a good brand given some reviewer comments.
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