[Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems

[Dan Perik]

Pardon my answering myself (and for the long post).  But I do have it
sort of working, and I come back with information on the GS HT-488, as
well as questions related to SIP / DTMF issues.

The GS HT-488 acts as a PSTN pass through device for 4 rings.  If the
phone attached to the FXS port hasn't picked up by 4 rings, it will by
default "answer", and you're at an internal (*) dial tone.  You can also
configure the HT-488 to dial a specific extention, which it will then do
instead of dropping you at an internal dial tone.  From there you can
obviously do what ever you want with the call.  (It would be nice if you
could configure and/or disable the # rings before it switches over to
VoIP.  Maybe that will be something they will add to a firmware update
someday.) 

For dialing out, you set up an extention for the FXO port, and dial
that.  It will ring once, and then present you with the PSTN line, dial
tone and all.   From there you (should be) are able to dial out. 

Now, here is my problem and question.  Both the FXS and FXO ports are
set up to use SIP INFO for DTMF.   You would think that when you have
dialed the FXO port, and are at the PSTN dial tone, the HT-488 will
translate the SIP DTMF INFO passed through to the FXO port as audible
DTMF on the PSTN line.  This is not the case.  So I really can't make
outgoing calls yet.  Now, I can change the FXS line to send DTMF in
audio, which works, but I figure that sending DTMF in audio is not
ideal.  So I'm trying to "translate" the SIP DTMF INFO to DTMF
in-audio.  I've tried a few combinations of SipDTMFMode(inband) (trying
to do a DTMF style translation, I guess), and
Dial(SIP/gs1-FXO,10,D(<PSTNnumber>) ), but can't get it to work.  

Should I just suck it up and keep the FXS port using DTMF in-audio, or
is there a way to get SIP DTMF INFO translated to DTMF tones in audio in
the Dial settings for the FXO extension?

Thanks!
Dan

Dan Perik wrote:

>I just got my shiny new Grandstream HandyTone-488 today.  My goal is to
>use it to allow incoming/outgoing calls to PSTN using my normal ole'
>phone as usual.  I will be switching over to using BroadVoice as my main
>phone #, but want that to be as seemless of a switchover as possible
>(for the wife and kids, and for people needing to call us).
>
>I've got the following working:
>
>FXS -> * ( and then -> BroadVoice )
>( BroadVoice -> ) * -> FXS
>FXO -> * ( and then -> FXS )
>
>I don't have this working:
>( FXS -> ) * -> FXO
>
>In other words, I can't seem to call out on my PSTN line from Asterisk.
><snip>
>  
>