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> > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
