Thanks. I see.
Regards. Scott On Wed, May 11, 2011 at 3:43 AM, John Novack <[email protected]>wrote: > Assuming you have read the link you provided, and understand most of what > it said, the link really doesn't address calling out over a POTS (copper) > line. > When Asterisk dials out and finishes the dial string, it considers it > answered. IF your POTS provider doesn't provide any clue, other than audio, > that the line is answered, not answered, or the call terminates, then you > will have to do some coding. > You could set an absolute limit, or IF the call will always go to you, you > could listen for some DTMF and hang up then. > OR, if there is an option, you could use some sort of digital trunk, SIP or > what have you, where there is more complete communication. > SIP isn't the most desirable, IMO, as some of your countrymen ( and other > counties s well ) seem to have nothing better to do than to attempt to break > in to VOIP systems and steal telephone time. > T1/E1 will certainly provide much better communication, as will ISDN. > > Remember the POTS analog technology was built and constantly modernized > over the last 130 years, but was never designed for anything other than > human communication. Once stupid machinery became involved, the problems > became larger and larger. > > John Novack > > > > Scott Zhang wrote: > > So does this mean no solution when used ZAP/DAHDI with PSTN line? > > If I installed an E1, will that work? > > > Thanks. > Regards. > > On Wed, May 11, 2011 at 12:57 AM, John Novack < > [email protected]> wrote: > >> Remember that ZAP/DAHDI channels don't receive ( because most PSTN/POTS >> lines don't provide ) answer supervision. >> This will certainly complicate what you want do do. >> >> John Novack >> >> >> Scott Zhang wrote: >> >> Hello. All. >> I am a bit new to asterisk, started from half a month ago. >> I am setting up a home asterisk server with analog card. I am using >> asterisk 1.4.27. >> At the moment, I bought a X100P card and installed it on my computer. >> I used it to connect my home phone line. For the moment, it works fine when >> dial in. Soon I noticed when I dial out through it to my mobile, it can't >> hang up automatically after I hang up my mobile. After googled, I found the >> reason as described as below link and some solutions. >> >> http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html >> For me, none of solutions works. >> So I am rethinking should I buy another TDM400P card. >> But I am wondering because in China. The phone system looks different >> so I don't know if TDM400P will work or not. >> >> Here is the flow when I am using X100P to dial out. >> 1. Pick up phone >> I hear tone. DA~~~ >> 2. press the number >> tone: DA~~~ >> 3. dialing~~~~ >> No more tone. Music playing~~~~~(lalala, I love lalal) >> At the same time, on asterisk console, it prints out. "The call has been >> answered". >> Actually it is still dialing and my mobile is ringing because I didn't >> answer the call.. The music was played by ISP >> 4. whether I answered the call or refuse the call. No more prints on >> asterisk console. >> But on phone end, when I refuse the call, instead of busytone, I hear the >> voice "The phone you're dialing is busy now. Please try again later.". >> So the whole thing is, during the whole call process, only before dialing, >> we can hear the phone tone, for all other time, Dialing, refused, the ISP >> will play music/voice instead of providing the tone. I don't understand how >> x100p identify the status, I guess should be on the tone. >> 5. I wait asterisk/x100p to hangup the call and after 5 minutes, I have to >> cut the phone line to force it hang up. >> >> So can TDM400X work with such a system without tone only with music and >> voice? >> >> Thanks. >> Regards. >> Scott >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> >> Dog is my Co-pilot >> >> > > -- > > Dog is my Co-pilot > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
