PSTN to Sipphone via SIpbroker?

[boatman]

Is it possible to call a Sipphone/Gizmo number via one of Sipbroker's PSTN access numbers? Does the Sipphone/Gizmo user first need to register with Voxalot or Sipbroker?

As yet, I have not been able to make the call, got a busy signal.

[DracoFelis]

Originally Posted by boatman

Is it possible to call a Sipphone/Gizmo number via one of Sipbroker's PSTN access numbers?

Yes, provided that the phone service you are using properly passes the tones to the PSTN number you are calling.

Originally Posted by boatman

Does the Sipphone/Gizmo user first need to register with Voxalot or Sipbroker?

No signup/registration needed. By default, SIPphone users should be just as reachable via SIP Broker, as they are from other SIPphone users.

Originally Posted by boatman

As yet, I have not been able to make the call, got a busy signal.

It might be your dialing sequence. After the voice prompt answers, you want to dial the sequence to reach Sipphone/Gizmo.

The proper dialing sequence for Sipphone/Gizmo is the SIP Broker provider code *747 followed by the full SIPphone/Gizmo dialing sequence (that one SIPphone user would use to call another). But what many people seem to forget/overlook, is that SIPphone itself requires 11-digit dialing sequences (for one SIPphone user to call another). So if (for example) you are trying to reach user 1-747-1234567 on SIPphone, you would have to use the SIP Broker dialing sequence of: *747 1 747 1234567

i.e. you can NOT use *747 and the 7-digit SIPphone number, as *747 just sends the call to SIPphone, and SIPphone requires 11-digits for the call. So you really do need a 15-digit (i.e. *747, followed by the 11-digit SIPphone dialing) sequence to reach SIPphone/Gizmo from SIP Broker (including the SIP Broker PSTN dial-in numbers).

[boatman]

Hey DracoFelis!

It's great to get advice from a famous guy like you! I have read many of your articles about VoIP.

I can call test numbers from my SPA2102 device which is registered with Voxalot, but same test number fails (slow busy tone) when I try it at a Sipbroker PSTN gateway. I have only IP phones here, my calls to the PSTN go out via carriers.icall.net. I have had some issues with my credit card company not always understanding my DTMF tones. The problem may be that my DTMF tones are not getting through to the Sipbroker PSTN gateway. I called the time at *393-612 as a test.

My SIP device is a Linksys SPA2102. There are some settings for DTMF stuff, here is what it's got now:

DTMF Process INFO: yes
DTMF Process AVT: yes
DTMF Tx Method: Auto
DTMF Tx Mode: Strict

Should I change any of those?

Thanks,
boatman

[DracoFelis]

Originally Posted by boatman

I have only IP phones here, my calls to the PSTN go out via carriers.icall.net. I have had some issues with my credit card company not always understanding my DTMF tones. The problem may be that my DTMF tones are not getting through to the Sipbroker PSTN gateway.

That sounds likely. If your phone company has issues properly sending DTMF tones, than pretty much all the PSTN gateways are going to have issues. Which will probably make it hard to "test" them.

OTOH if you have IP phones, why not just use either sipbroker.com or voxalot.com directly (bypassing the PSTN numbers)? Not only will that likely be more reliable, but it will also avoid tying up one of the PSTN numbers for your call. i.e. accessing the sipbroker.com proxy directly, is not only more likely to work from your IP phones, but it's also more efficient routing on the SIP Broker end of things as well (making it a win-win situation IMHO).

Originally Posted by boatman

My SIP device is a Linksys SPA2102. There are some settings for DTMF stuff, here is what it's got now:

DTMF Process INFO: yes
DTMF Process AVT: yes
DTMF Tx Method: Auto
DTMF Tx Mode: Strict

Should I change any of those?

DTMF with VoIP is always a YMMV thing. Sometimes you can never get it to work, and at other times one or more of the settings work (and which settings work with DTMF also tends to vary with the provider you are using, making it also a YMMV thing).

What CODEC do you use for your VoIP? It wouldn't be the high bandwidth G711u CODEC by any chance? If so, try setting the DTMF transmit method to "InBand", as that often (but not always) works best with the G711u CODEC. However, keep in mind that "InBand" will ONLY WORK with the two high bandwidth CODECs (i.e. G711u and G711a). For all other CODECs you will get total DTMF failure if/when you try "InBand"...

[boatman]

I have only IP phones here. The only reason I wanted to dial the gateway was to test that it worked before giving the number to someone near that gateway.

I use the G711u codec. I would like to study up on what all those different DTMF settings mean, but I can't find the technical docs for the SPA2102, Linksys seems to have abbreviated the user manual for this model. If anyone can point me to the full docs for the SPA2102, that would be super.

I have been having a little trouble with DTMF when calling my credit card company's automated info service, so my DTMF does need some tweeking in order to work better.

[DracoFelis]

Originally Posted by boatman

I have only IP phones here. The only reason I wanted to dial the gateway was to test that it worked before giving the number to someone near that gateway.

But that's the problem with testing from a VoIP phone. I would suggest making the test call from either a friend's "land line", or from a cell phone.

Originally Posted by boatman

I use the G711u codec.

Then give "InBand" a try, and see how it works for you. While InBand is technically the least reliable of the options, it's also the option that is most universally supported (assuming you use the G711u CODEC).

BTW: What "InBand" really does is send the tones as if they were just some other "voice" on your VoIP line. This means that if your VoIP can properly send this "voice" sufficiently undistorted (which is why the less distorting, but high bandwidth, G711u CODEC is needed), than the tones WILL get to the other end no matter what tone support is available on the other end. However, for the same reason this is the least reliable (albeit the most universally "supported") of the options, as anything that distorts the tonal quality of your "voice" will distort the tones (and cause problems with them on the other end).

Originally Posted by boatman

I would like to study up on what all those different DTMF settings mean, but I can't find the technical docs for the SPA2102, Linksys seems to have abbreviated the user manual for this model. If anyone can point me to the full docs for the SPA2102, that would be super.

If you are willing to go with an older PDF manual for a slightly different (but closely related) model adapter, check out the downloadable manual on the Sipura web site (these adapters used to be made by Sipura, before LinkSys bought them out):

http://www.sipura.com/Documents/Sipu...uidev2.0.9.pdf
http://www.sipura.com/support/index.htm

BTW: From the above listed manual, here is the short versions of what the DTMF settings mean/do:
Method to transmit DTMF signals to the far end:
Inband = Send DTMF using the audio path;
INFO = Use the SIP INFO method,
AVT = Send DTMF as AVT events;
Auto = Use Inband or AVT based on outcome of codec negotiation

FYI: I used to have my adapter set to "Auto", but changed it to "InBand" after discovering that my primary VoIP provider's voice mail didn't work with "Auto" (but did with InBand explicitly forced). Go figure...

[vikasintl]

Originally Posted by DracoFelis

Yes, provided that the phone service you are using properly passes the tones to the PSTN number you are calling.


No signup/registration needed. By default, SIPphone users should be just as reachable via SIP Broker, as they are from other SIPphone users.


It might be your dialing sequence. After the voice prompt answers, you want to dial the sequence to reach Sipphone/Gizmo.

The proper dialing sequence for Sipphone/Gizmo is the SIP Broker provider code *747 followed by the full SIPphone/Gizmo dialing sequence (that one SIPphone user would use to call another). But what many people seem to forget/overlook, is that SIPphone itself requires 11-digit dialing sequences (for one SIPphone user to call another). So if (for example) you are trying to reach user 1-747-1234567 on SIPphone, you would have to use the SIP Broker dialing sequence of: *747 1 747 1234567

i.e. you can NOT use *747 and the 7-digit SIPphone number, as *747 just sends the call to SIPphone, and SIPphone requires 11-digits for the call. So you really do need a 15-digit (i.e. *747, followed by the 11-digit SIPphone dialing) sequence to reach SIPphone/Gizmo from SIP Broker (including the SIP Broker PSTN dial-in numbers).

I was trying my voxalot number through pstn numbers via calling through icall.com software and after voice prompt I was entering *010-my voxalot number" am I doing something wrong?
surprisingly when I enter enum number it rings my x-lite sofware but now when I enter *010-voxalot number...let me know if am doing someghing wrong or please explain after adding sip code which is *010 what should I enter ?