Author Topic: Asterisk users - any up to date list of pulse capable ATA's? (Read 7558 times)

mdodds · « **Reply #15 on:** April 16, 2015, 11:32:13 AM »

Sorry unbeldi, you are correct. It's the server guy in me coming out

I consider anything that's not LTS to be a dev release, such as any Ubuntu release that's not an even year .04 version, even though I know they are really not dev versions. On a related note, have you used Asterisk 13?

mdodds · « **Reply #16 on:** April 24, 2015, 02:20:05 PM »

Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee

I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?

unbeldi · « **Reply #17 on:** April 24, 2015, 02:24:14 PM »

Quote from: mdodds on April 24, 2015, 02:20:05 PM

Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee
I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?

It should work. What is the firmware version on yours?

Software Version: Program-- 1.0.5.10 Bootloader-- 1.0.0.15 Core-- 1.0.5.9 Base-- 1.0.5.10

This is not the latest version however, which I think is 1.0.13 or so.

mdodds · « **Reply #18 on:** April 24, 2015, 02:40:35 PM »

Here are the versions:
HT-502 V2.0A
Software Version: Program -- 1.0.12.1 Bootloader -- 1.0.0.18 Core -- 1.0.12.1 Base -- 1.0.12.1
Extra -- 1.0.12.1 CPE -- 1.0.1.44

unbeldi · « **Reply #19 on:** April 24, 2015, 02:55:35 PM »

Mine is
Product Model: HT-502 V1.2A

unbeldi · « **Reply #20 on:** April 24, 2015, 03:08:44 PM »

Set the Preferred DTMF mode to SIP INFO, and disable DTMF negotiation.
Establish a call to the unit from another extension, or use a DTMF tone dialer with the rotary phone to call another extension.
While maintaining the call, watch the SIP messages when dialing a number on the phone during the call. You should see SIP INFO messages with the digit that was decoded from the dial action.

For example you should see a SIP message like this when dialing "6":

INFO sip:0666@10.201.239.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.239.171:5060;branch=z9hG4bK492069190;rport
From: "5021" <sip:5021@pia.example.org>;tag=1173897288
To: <sip:0666@pia.example.org>;tag=as6b896e56
Call-ID: 551528776-5060-10@BJJ.CAB.CDJ.BHB
CSeq: 104 INFO
Contact: <sip:5021@10.201.239.171:5060>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-502 V1.2A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=240

markosjal · « **Reply #21 on:** December 08, 2016, 09:48:38 PM »

ALL of my calls are VoIP in one form or another, no matter where I am and many originate from Rotary dial phones. Luckily the 2 cities I dial most have no prefix overap so I have 7 digit dialing to fixed lines in 2 cities . There are some identical prefixes but that is where one is a cell phone and in Mexico cell phones are never dialed the same as fixed phones so it is a non-issue. (there are some cell phones in Mexixco with fixed dialing however)

I run X86 Vanilla Asterisk at home as a Cellular gateway using Chan_dongle and POTS gateway. This system is in my home and I do not register phones to it. It recives calls voia POTS and Cell (with Fixed dialing number) and sends to main server. Also handles many outbound calls

I have another Cellular Gateway running Vanilla Asterisk on X86 in city 2 no phones registered to it. it receives calls via incoming (fixed dialing) number and routes them to main server

I had a Raspberry Pi running Free PBX as a cellular gateway in a third city. No phones registered to it. This is now off line

I bring it all together on a Elastix server running as a VPS on a dedicated server which I rent and divide into PBXs on VMs for clients. One of those PBXs is my own

for rotary dial phones I have found no better ATA than the UTStarcom IAN-02EX . I have tried the grandstream HT503 (only very recently) but have issues with some rotary phones, particularly Ericssons. This is not an issue of dial speed rather a misread number with an extra pulse. I have unlocked Vonage Vportal only to find that it is very unreliable and often goes unregistered saying "cannot connect to Vonage network (rather my server)". I was told this may be from using both ports but it still seems the same with only one port. Although the Vportal does seem to accurately read the digits consistently, and displays them on display.

If you do look into a UTStarcom IAN-02EX (about 28 USD NEW here in Mexico) absolutely avoid unlocked units as you may only be able to use one port. I have used MANY rotary phones on these models and never had an issue , even when I had questionable dial speed.

So yes I have asterisk between my rotary phones and POTS lines as well as between my rotary phones and VoIP liines and between rotary phones and cell lines (that's a rare one).

I also use :
Google voice with Simonics.com as a go-between for USA number and outbound calling
FreePhoneline.ca for C anadian number and calling
messagenet.it for a UK number
VoIP.ms additional ported USA numbers and outbound calling
Rapidvox outbound provider
Flynumber.com for an Australian number, and Canadian number

All with Least Cost Routing because I am cheap (Cheap is good free is better, so Free is a very good price!)

NorthernTelcom if you need help with your configuration , call me
Mark
1 503 489 7870 USA
1 604 800 9818 Canada (soon to be migrated to a new number and not sure what that is at the moment)

timmerk · « **Reply #22 on:** January 30, 2017, 03:17:32 PM »

Quote from: markosjal on December 08, 2016, 09:48:38 PM

If you do look into a UTStarcom IAN-02EX (about 28 USD NEW here in Mexico) absolutely avoid unlocked units as you may only be able to use one port.

Did you mean avoid *locked* units, not unlocked?

markosjal · « **Reply #23 on:** August 16, 2017, 05:12:52 AM »

What I mean is any unit that was locked to a provider and is now "unlocked" . The UTStarcom from Lingo had bad firmware and only one port was usable.

andy1702 · « **Reply #24 on:** August 17, 2017, 03:49:05 AM »

Quote from: mdodds on April 24, 2015, 02:20:05 PM

Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee
I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?

I use a Grandstream 502 and it works with pulse dialling. The first thing I did when I got it was to turn off any firmware updates in the settings. I've had it about 18 months now. The only problem I had was that it didn't work with certain UK phones. It turns out the make/break ratio of the UK dials can fall just outside the range the Grandstream will allow. A quick tweak of the dial contacts cures this though.

mdodds · « **Reply #25 on:** August 20, 2017, 11:56:56 AM »

Hi all, I'm back after a fairly lengthy hiatus.
There seems to be 2 main hardware revisions for the 502, 1.2 and 2.0. I have not had any luck with 2.0 versions and pulse dialing. Unfortunately 1.2's are only available used. I did have success with my Digium IAXy ATA, but I only have one of those and they are discontinued now.
However, my Panasonic KX-T61610 just arrived yesterday so the pulse to DTMF problem is pretty much solved!

andy1702 · « **Reply #26 on:** August 21, 2017, 03:00:05 PM »

The Panassonic is definitelty the way to go. It's much more forgiving of slightly mis-set dial speeds etc than the Grandstream ATA.

For Grandstream users, I wonder if there is a way of rolling back the firmware?

markosjal · « **Reply #27 on:** August 27, 2017, 10:15:52 AM »

Quote from: andy1702 on August 17, 2017, 03:49:05 AM

I use a Grandstream 502 and it works with pulse dialling. The first thing I did when I got it was to turn off any firmware updates in the settings. I've had it about 18 months now. The only problem I had was that it didn't work with certain UK phones. It turns out the make/break ratio of the UK dials can fall just outside the range the Grandstream will allow. A quick tweak of the dial contacts cures this though.

Similar to what I have seen with some Ericsson dials from Mexico as well. It is not the pulses per second it seems rather the duration of each pulse. Much depends on the model/age of the dial it is the newer ones that seem to be problematic, not the older ones.

Payphone installer · « **Reply #28 on:** August 27, 2017, 10:32:42 AM »

http://www.dslreports.com/forum/r25525579-Looking-for-an-FXO-device-needs-pulse-dialing-support

kb3pxr · « **Reply #29 on:** December 23, 2017, 12:19:14 PM »

I'm researching this as well, here is what I found. All Devices are Grandstream and this is not a complete list.

HT502
HT70X (All in this series with updated firmware.
HT80X/81X with firmware 1.0.5.11 (New December 2017) or higher

The Following settings need changed on the ATA (dial plan assumes PBX):

High Ring Power to Yes: This increases the available ring current
Enable Pulse Dial to Yes: This enables the Rotary Dial (you can leave this off for touch tone).
Dial Plan (Pulse/15xx Only): add the following in the front for every feature code with a * <11FC=*FC> | Where FC=digits of feature code. This will get complicated for speed dial and parking as speed dial would be <110SPD=*0SPD> |where SPD = Speed dial number (the entire dialed code must be specified). Also Remember the "|" as that is the OR keyword.

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