If you use asterisk or freeswitch you should know......

[markosjal]

A little tidbit of useful informaton that contrary to poular belief there are ATAs that support rotary dialing ...

One that i used (until it gave up recently) is The UTStarcom IAN-02EX . I have purchased these in the past and actually have a source for them now here in Mexico. You can find them used on Ebay from time to time just make sure you do not get a locked unit.

On a further note ....
Custom contexts allow you to prefix a dialing string in asterisk . So lets say I have to dial *43 to get to the echo test I can prefix any 2 digit dialing sequence with * . So when I dial 43 asterisk translates that to *43. When I dial 99 asterisk would translate that to *99.

I do not believe it will do true DTMF conversion like tone dialing when you reach an IVR; press 1 for sales 2 for support.

It does however allow you to use your antique phones for everyday use with VoIP , without a convoluted solution .

Mark

[andy1702]

The Grandstream 502 supports pulse dialling too. The only thing is your dial's speed has to be absolutely spot on 10pps. Any slower and it won't work or will misdial. A little faster will work, but not much faster.

[markosjal]

There were touch pulse pads that generated 20 pps as I recall . This is probably the reason why the HT502 will take faster .

Personally I have an HT503 and after reading posts here I fired it up with a rotary phone and sure enough it worked. Some say their HT503s  have not , so could be firmware related.

I do not like the Grandstreams much.  The reason I shelved the HT503 is I did not like the sound quality of the FXO nor the FXS. I always got a buzz on it aside from voice quality issues. I have read in VOIP forums that others have experienced the same and the buzz gets worse over time. This problem is exasperated by the compression used in some modern cordless phones as the buzzing gets increased by the compression during silence. The buzzing is more tolerable on old phones, and almost seems "right" to have some noise on them!

The Chinese clone FXO in my gateway does the job for my analog line better than anything else.

A Cisco/Linksys FXO burned up after only one year.

Maybe I am spoiled by the sound quality of the Mediatrix and other "good" ATAs but I put the UTStarcom up there with the best of them . I am expecting a new one tomorrow to get more rotary phone ports.

Also I hacked a Vonage Vpotal to my asterisk, for use with rotary phones, and resurfaced the front removing the Vonage logo. It looked quite nice next to the old black Ericsson and served as a caller ID box and worked with pulse dialing, but I find it to be quite unreliable.   It seems I make a call on one line and BOTH lines go unregistered for a while after the call . Vportal says "Your vonage device is not connected to the network". No amount of tweaking seems to resolve it. Asterisk complains the the peerpoke has exceded 2000 ms at this time to but even disabling qualify does not help much. If it worked reliably, it would be my device of choice.

Mark

[unbeldi]

I disagree with the statement that the HT502 has a very narrow acceptance range for dial pulse speed.
By my observations, the device starts failing below ca. 8.5 pulses per second.  Most Bell specs called for adjustment starting about that value. Unfortunately I can't locate my exact measurements with standard telco test equipment anymore for the HT502, but 8.5 PPS is better than what a modern Verizon FIOS line can do, which fails below 9 PPS.

Grandstream does not officially "support" pulse dialing, it is not stated in any of their manuals, so we should state that the HT502 and 503 accept it, not that they support it.
I have heard of reports where a firmware upgrade has removed the pulse capability, but I have not seen proof of that. I never updated any beyond 1.0.5.10.

The Vonage Vportal, VDV21, is actually a very nice unit. I disagree that it is not "stable", but one has to live with the algorithms it uses to maintain "state" with the Vonage servers. It is best to remove all Vonage IP addresses from the firmware image that are used to transmit telemetry info to Vonage. This requires some detective surgery on the firmware image. I replace them with generic hostnames and/or addresses on my network, so they don't "call home" any more.  The units do reboot automatically on a schedule, but only when no call is ongoing at that time. Other Vonage ATAs do that too, it is not just the VDV series. I have used these for years now without problems.  One nice aspect is that not only is their pulse support quite good, the units can also be configured for specific ringing frequencies other then 20 Hz, via the command line interface.  Of course having an integral caller ID display on a SIP ATA is another novelty feature.  There are millions of them available, and one should not pay more than $5 to $10 to get one delivered.

[markosjal]

I did much improve my VDV 21 by adding outbound_proxy and outbound_proxyport settings . It still seems to disconnect several times a day and it has seemed to reboot itself in the middle of calls but not so much now (once in the last few days).