[asterisk-users] About X100P and TDM400P analog card in China

[Scott Zhang]

Thanks.
I see.


Regards.
Scott

On Wed, May 11, 2011 at 3:43 AM, John Novack
<jnovack at stromberg-carlson.org>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 <
> jnovack at stromberg-carlson.org> 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
>>
>>
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>>
>
> --
>
> Dog is my Co-pilot
>
>
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