the Google Voice dial tone

[dialtone]

This is SP1:
*  SP1 is connected to a Google Voice account, number 123-456-7890
*  SP1 -> CallForwardUnconditionalNumber -> SP2(9876)
*  All inbound calls to the Google Voice number 123-456-7890 make the SIP 9876 number phone ring.

This is SP2:
*  SP2 is registered to a Generic SIP account, number 4567
*  SP2 -> CallForwardUnconditionalNumber -> SP1(1234567890)
*  All calls to the SIP number 4567 provide access to the Google Voice menu for the Google voice number 123-456-7890

The AA for the OBi:
*  Auto Attendant -> PrimaryLine -> SP1 Service
*  Therefore the Phone on the PHONE port will use the Google Voice service for all outbound calls.

In other words, just pick up the Phone on the PHONE port you get the Google Voice dial tone.
So it works exactly as expected.

Similarly, it is possible to directly access the Google Voice dial tone by changing the "1234567890" in
SP2 -> CallForwardUnconditionalNumber -> SP1(1234567890)
without going to the Google Voice menu each time you want to dial out?

What would that change be?

[RonR]

I've started to respond to your post a couple of times, but each additional time I read it, I just get more and more dizzy.   

What exactly is it you're trying to accomplsih?

[RonR]

In other words, just pick up the Phone on the PHONE port you get the Google Voice dial tone.

FWIW ...

1. Any time you pick up the phone connected to the PHONE Port on the OBi, you're listening to a dialtone generated by the OBi.

2. Google Voice and VoIP providers don't provide dialtone.  It's generated by OBi's, ATA's, hardphones, softphones, etc.

[dialtone]

I've started to respond to your post a couple of times, but each additional time I read it, I just get more and more dizzy.   

Sorry about the confusion.
I will try to be as clear as I can.

[dialtone]

In other words, just pick up the Phone on the PHONE port you get the Google Voice dial tone.

FWIW ...

1. Any time you pick up the phone connected to the PHONE Port on the OBi, you're listening to a dialtone generated by the OBi.

2. Google Voice and VoIP providers don't provide dialtone.  It's generated by OBi's, ATA's, hardphones, softphones, etc.

Understood.  Agreed.
All I meant to say that - I want to hear that dial tone (generated by the OBi).

[dialtone]

What exactly is it you're trying to accomplsih?

SP1 is connected to a Google Voice account, number 123-456-7890
SP1 -> CallForwardUnconditionalNumber -> SP2(9876)
Where 9876 is an extension on an Asterisk system.
There are 4 extensions on that Asterisk system.

SP2 is SIP account registered to the same Asterisk, number 4567
SP2 -> CallForwardUnconditionalNumber -> SP1(1234567890)

On that Asterisk system when anybody dials 4567, currently it takes directly them to the Google Voice menu ...
... and almost always they dial 2 then nnn-nnn-nnnn# to make a call to any US number.

Can they skip this Google Voice menu entirely?
So from any Asterisk extension they dial 4567 (hear the OBi generated dial tone) then dial nnn-nnn-nnnn#

[RonR]

I'll bow out and let someone more familar with Asterisk give you advice on this one (but I've got to believe there's a much simpler approach than the one you're currently taking).  Please don't take offense, but this looks like a Rube Goldberg machine: dialing out through a VoIP provider to the PSTN to call into a Google Voice PSTN number to then have Google Voice place an outgoing PSTN call.  I hope I'm still missing something.   

[QBZappy]

dialtone,

Are you certain option #2 you mentioned is Google voice. Option #2 in Obi is to make a new call. Is the IVR voice male or female?

[dialtone]

dialtone,

Are you certain option #2 you mentioned is Google voice. Option #2 in Obi is to make a new call. Is the IVR voice male or female?

Yes, it is Google Voice.
It is a female voice saying, "You have no new messages. To place a call press 2. Or to change your settings press 4."

[QBZappy]

dialtone,

Can they skip this Google Voice menu entirely?
So from any Asterisk extension they dial 4567 (hear the OBi generated dial tone) then dial nnn-nnn-nnnn#

The above statement sounds like your stated objective. The reasons behind your current setup strategy of unconditional CF is not clear.

You have not given too many details about your Asterisk setup as well. If you want to achieve your stated objective of hearing OBi dial tone, you need to set it up as a trunk on the Asterisk. Also setup "ITSP InboundCallRoute" to something like this {Asterisk_Extension(4567)>(xx.):sp1},

Incoming:
At present all GV calls are sent to Asterisk Ext#9876
Outgoing:
Is setup to dial your own GV number that is why you get GV IVR

I don't think you need to hear the OBi generated dial tone. What you need to hear is the Asterisk generated dial tone. You seem to be trying to make an outgoing call over the OBi via SP1 (which is a GV channel).

You need to change you Asterisk setup to accomplish this. Some of the Asterisk setup instructions found here might give you some insight. Have a look.

Setup Asterisk (FreePBX) with Obi to use the AA features of the Obi
http://www.obitalk.com/forum/index.php?topic=68.msg155#msg155

[MichiganTelephone]

I think part of the reason for the confusion here is that unless I'm wrong, the OBi device will only send dial tone to the Phone port. It's not designed to provide dial tone to anyone coming in over a Service Provider or OBiTALK network channel.

What I don't understand is why you're trying to force users to do two stage dialing in the first place.  Just let them dial the number they want, and let Asterisk process it and send it to Google Voice via your OBi device.  See How to use the Obihai OBi100 or OBi110 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network (note that these instructions are rather specific to a setup that includes FreePBX, but even if you are running bare Asterisk you should be able to use the same principles outlined in that article, particularly with respect to how you configure your OBi device).

If for some reason you really want them to hear a second dial tone, the best place to provide that is in their endpoints, but not all endpoints have the capability to provide a second dial tone (if they can do it, you usually have to program it into their dial plan or digit map).  Sending touch tone digits over a VoIP connection is not always 100% reliable, depending on several factors such as the codec(s) used in the call path.  It's much better to have your users dial the number they want to call (with or without a routing prefix), send the entire number from their endpoint to Asterisk, and let Asterisk figure out how to route it.