[asterisk-dev] usbb3g progress
Sam Liddicott
sam at liddicott.com
Fri Feb 27 09:37:13 CST 2009
I think (looking at chan_misdn.c) that I have to generate the tones myself.
Sam
* Sam Liddicott wrote, On 27/02/09 14:53:
> I now pretty much have it all sewn up, these final questions may be user
> questions rather than dev questions; if so, please forgive me.
>
> I can dial a number to get the FXS side of the USBB3G which gives me a
> dial tone on the PSTN. If my IP handset uses DTMF dialing - then I can
> dial - otherwise I can't. What sort of tricks are required to get
> asterisk to generate DTMF for RFC#### dialing handsets?
>
> Ideally I would like to not even get the PSTN dial tone but have
> asterisk do the dialing for me. Do I need to do anything to handle these
> cases in my channel driver, (like looking at and parsing "dest" and then
> persuading asterisk to emit the tones) or is it a matter of
> extensions.conf and the like?
>
> Finally (and this is technical) macros like this:
> exten => incoming,1,Answer()
> exten => incoming,n(exten),Background(vm-enter-num-to-call)
> exten => incoming,n,WaitExten(5)
>
> work when called from an IP phone, but when dialed in on the analog line
> the audio is very garbled - but yet the audio is fine if an IP phone
> answers the analog line instead of these macros, so I'm not sure why the
> audio would be garbled only for the macros.
>
> I'm guessing it is two-way garbled because when I press DTMF buttons I
> don't get connected.
>
> Sam
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
More information about the asterisk-dev
mailing list