[EMAIL PROTECTED] wrote: > Ahh, I see. Is this true of all Channel Banks connected through T1? I > am less concerned about fixing my testing environment as in making > sure the program works in as many customer environments as possible. > If I have to rule out this fairly common environment, it would be too > bad (though not the end of the world). >
Honestly, I am not 100% sure if this is generally true for all channel banks but in my experience, I yet have to see an FXO device that handles call answer supervision properly. > I definitely don't want to go the DSP route: > http://www.3amsystems.com/wireline/tone-search.htm scares me. > You need not go this path. I am sure there are open source DTMF decoders lying around. Here's a good example http://cvsup.pt.freebsd.org/cgi-bin/cvsweb/cvsweb.cgi/src/usr.sbin/i4b/dtmfdecode/?only_with_tag=MAIN#dirlist > > > joegen-at-opensipstack.org joegen-at-opensipstack.org |SIP > Implementors| wrote: >> Bruce, >> >> It's not not asterisk. T1->Channel Bank does produce an early seize >> signal. Your best bet would be to use a tone analyzer to detect the >> ring back in-band in your announcement UA. This will be quite a >> daunting task because you will need to deploy different sets of tone >> detection for the different frequency used in different countries. >> My advise, get rid of the channel bank and use T1 all the way. >> >> Joegen Baclor >> >> Bruce Atherton wrote: >>> Actually, my setup is a Sangoma T1 card connected to an Adtran 750 >>> channel bank with some FXO ports. But you are right that the whole >>> pathway is controlled by Asterisk. >>> >>> Dale.Worley wrote: >>> >>>> From: Bruce Atherton <[EMAIL PROTECTED]> >>>> >>>> That is really good news. So the problem must just be with >>>> Asterisk, which happens to be the PBX I am testing with. I will >>>> take the issue up with the developers there. >>>> >>>> Oops, my mistake -- IIRC, you might be using a Digium PSTN interface >>>> board, plugged into the Asterisk host. In which case, everything >>>> regarding the PSTN interface is driven by the Asterisk code. But I >>>> assume the Asterisk developers know all about that, and any >>>> idiosyncrasies it has. >>>> >>>> Dale >>>> _______________________________________________ >>>> Sip-implementors mailing list >>>> [email protected] >>>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >>>> >>> >>> _______________________________________________ >>> Sip-implementors mailing list >>> [email protected] >>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >>> >>> >>> >> > > > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
