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Asterisk users - any up to date list of pulse capable ATA's?

Started by mdodds, April 14, 2015, 09:12:55 AM

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mdodds

I run an Asterisk server and I was wondering if anyone has an up to date list of current ATA hardware that supports pulse dialing.
I have a Grandstream HT-486 (doesn't work on pulse), an HT-503 (also doesn't work), and a Digium IAXy (works!).

I would like to be able to hook up my phones 1 to 1, or at worst 2 to 1 on ATA's. I have read that pulse works on the Grandstream HT-502, but I have also read it works with the 503, which it doesn't. Unfortunately the Digium IAXy (S101) isn't made anymore.
When I search for pulse compatibility it seems all the results are 3 - 5 years old and we know what 3 - 5 years means in the VoIP world :)

Jack Ryan

The Grandstream HT-502 accepts pulse dialling but the newer devices don't. I don't know of any other new devices that support pulse dialling and I don'really expect any to be introduced.

Jack

NorthernElectric

Are you using your asterisk between your dial phones and a POTS line?  If so, care to share your extensions.conf?
Cliff

Weco355aman

If your P.C. has a PCI slot, you could buy a 1 or 2 port Ds1 card (t1) card and a Adit 600 with FSX cards. This would provide 24 Stations TT/Dp. On Ebay you  can find the t1 card for about $85-125 and the adit 600 for $10-$20. I've bought fully loaded Adit 600 for $10 + $18 shipping.
This will provide 100 x 's better results that the ATA's you find.
This may be more involved than you want but a idea.
Seach for T1 under Asterisk.

Phil

mdodds

Cliff,
No more POTS lines here at all, all SIP trunks. Are you using "pure" Asterisk or something like FreePBX?

mdodds

No PCI slot, or any slots :) I migrated my * server to a Raspberry pi about a year ago. I could accomplish what you're talking about with a SIP to T1 gateway like I installed a couple months ago for a customer, but those things are still a bit pricey! As luck would have it, I have to order a 2 port ATA for a customer, so I'll at least be able to test a 502 and make sure it works :) You'd think a 503 would work if a 502 does, but mine doesn't :(

NorthernElectric

Quote from: mdodds on April 14, 2015, 04:24:36 PM
Cliff,
No more POTS lines here at all, all SIP trunks. Are you using "pure" Asterisk or something like FreePBX?

I'm running "pure" Asterisk, v. 1.8.32.2 on FreeBSD 10.1.

I had a working setup a few years ago just using a Motorola chipset based modem as a clone X100P FXO and a soft phone on my Windows PC.  I was mainly using it as an answering machine, but I think I had it configured so that I could answer incoming calls with the soft phone, and possibly so that I could make outgoing calls too.  For some reason, I shelved it.  I thought I still had the setup but I must have reused the hard drive for another project.

This time around, I bought a used Digium TDM400P with an FXO module included.  I plan to get up to 3 FXS modules to plug vintage phones into.  I am under the impression that I should be able to support pulse dialing with this hardware.  I'll probably start out with 1 and see how it goes.

I have a number of ideas on things I'd like to try with it, like capturing the 10 digit number, deciding if it's long distance or not, and if long distance connect to my calling card access number and place the call through it, otherwise just make a local call.  Of course I will get the voicemail feature going again.  I have recorded an outgoing message with a SIT tone at the beginning hoping to fool autodialers into thinking my line is out of service.  I tried this idea briefly in my previous asterisk implementation but I don't think I used it long enough to determine if it really worked.  I'm also thinking of trying to interface a dial-less D1 or candlestick to the audio ports on my Windows system to use as input/output to the soft phone.

I might not actually get much of this done until next winter though, as the transition from winter to spring is progressing here and I will probably soon be spending more of my spare time outdoors.   :)
Cliff

mdodds

I bet you'd find that using one of the Asterisk distros that use FreePBX will be a LOT easier, particularly with some of the fancier stuff you want to do. Asterisk is up to version13 now (which is actually 1.13) and if you were using zaptel for your analog cards it has now gone away too, replaced by DAHDI.
FreePBX makes it a cakewalk to do things like you were talking about with call routing depending on the number you dialled etc.
There is PBXinaFlash, Elastix, AsteriskNOW and even FreePBX themselves have a native distro.
As far as I know the Digium TDM cards handle pulse dialling just fine.

NorthernElectric

Quote from: mdodds on April 14, 2015, 10:46:17 PM
I bet you'd find that using one of the Asterisk distros that use FreePBX will be a LOT easier, particularly with some of the fancier stuff you want to do. Asterisk is up to version13 now (which is actually 1.13) and if you were using zaptel for your analog cards it has now gone away too, replaced by DAHDI.
FreePBX makes it a cakewalk to do things like you were talking about with call routing depending on the number you dialled etc.
There is PBXinaFlash, Elastix, AsteriskNOW and even FreePBX themselves have a native distro.
As far as I know the Digium TDM cards handle pulse dialling just fine.

I am running mine on older hardware so prefer a minimalistic OS installation with only the packages I need to get the job done.  I haven't even put X on it and probably won't.  I am using asterisk 1.8.32 because that version is available as a binary package for FreeBSD 10.1.  It also has a binary package for dahdi 2.4 so I'm using that.  Asterisk 1.8.32.2 was released on Jan. 28 2015 so it's only about 2-1/2 months old.  It looks like they have changed their version numbering scheme so that every little bug fix jumps a major number.  If these versions work for me I see no reason to update them from source, but I will if I need to.  As for asking to see other's configs, I probably don't really need help, just thought it would save me a bit of the time I would otherwise spend figuring it out.
Cliff

mdodds

Actually they have been fairly consistent with their versioning except for that jump from 1.8 to 10....not sure whatever happened to 1.9 :) Just add 1. in front of their current releases and you get the REAL version. One thing to note is that 1.8x goes EOL this October.
What type of hardware are you running it on?

NorthernElectric

Quote from: mdodds on April 15, 2015, 02:56:04 PMWhat type of hardware are you running it on?

It's a Pentium-M 1.86 ghz on a socket 479 mini-ITX motherboard.  I chose this setup for the lower power consumption and thermal characteristics such that I could run without a cpu fan.  I got the biggest solid copper heatsink I could find that fit in the case.  I'm not crazy about the case, but I needed one that had one full height PCI slot and didn't need a riser card for a sideways slot.  I'm thinking maybe next winter I will 'update' it with a Core 2 duo mobile.   :)
Cliff

unbeldi

Quote from: mdodds on April 15, 2015, 02:56:04 PM
Actually they have been fairly consistent with their versioning except for that jump from 1.8 to 10....not sure whatever happened to 1.9 :)

That is also consistent with prior habits.  Until the time of the jump to version 10, the odd numbered versions were development versions in the source code control system.

Released were only the even versions, and upon a release the version was incremented for the new development branch.
Therefore 1.9 was the development branch for 10.

The reason for the change in numbering was that there were some discussions whether Asterisk could possibly every morph into a version 2 system and what it possibly could mean in terms of features.  If I recall, it was Marc S. who felt that this would never happen, and made the decision to remove the possibility.

mdodds

And for some reason they morphed into using the odd versions for the preferred releases (LTS), and even versions are dev releases.
They can spin it however they want, but I still think the jump from 1.x to 10, 11 etc. is a marketing ploy :)

mdodds

Cliff,
Most of mine are running either on raspberry pi's or 1.86 GHz Atom boxes. The Atom is perfectly happy up to 20-25 conversations as long as you don't transcode, and most of my customers need far less than that.

unbeldi

Quote from: mdodds on April 15, 2015, 09:21:51 PM
And for some reason they morphed into using the odd versions for the preferred releases (LTS), and even versions are dev releases.
They can spin it however they want, but I still think the jump from 1.x to 10, 11 etc. is a marketing ploy :)

It is incorrect to characterize the even numbered versions as development versions. They are standard releases but not covered by long-term support.  The odd-numbered development versions prior to 10 were never released, they were only available as snapshots, obtained directly as checkout from the source control system.