Thanks. <br>I see. <br><br><br>Regards.<br>Scott<br><br><div class="gmail_quote">On Wed, May 11, 2011 at 3:43 AM, John Novack <span dir="ltr"><<a href="mailto:jnovack@stromberg-carlson.org">jnovack@stromberg-carlson.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff" text="#000000">
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.<br>
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.<br>
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.<br>
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.<br>
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.<br>
T1/E1 will certainly provide much better communication, as will ISDN.<br>
<br>
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. <br><div><div></div><div class="h5">
<br>
John Novack<br>
<br>
<br>
<br>
Scott Zhang wrote:
<blockquote type="cite">So does this mean no solution when used ZAP/DAHDI with
PSTN line?<br>
<br>
If I installed an E1, will that work?<br>
<br>
<br>
Thanks.<br>
Regards.<br>
<br>
<div class="gmail_quote">On Wed, May 11, 2011 at 12:57 AM, John
Novack <span dir="ltr"><<a href="mailto:jnovack@stromberg-carlson.org" target="_blank">jnovack@stromberg-carlson.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">
<div bgcolor="#ffffff" text="#000000">
Remember that ZAP/DAHDI channels don't receive ( because most PSTN/POTS
lines don't provide ) answer supervision.<br>
This will certainly complicate what you want do do.<br>
<br>
John Novack<br>
<br>
<br>
Scott Zhang wrote:
<blockquote type="cite">
<div>
<div>Hello. All.<br>
I am a bit new to asterisk, started from half a month ago. <br>
I am setting up a home asterisk server with analog card. I am using
asterisk 1.4.27. <br>
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.<br>
<a href="http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html" target="_blank">http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html</a><br>
For me, none of solutions works. <br>
So I am rethinking should I buy another TDM400P card. <br>
But I am wondering because in China. The phone system looks
different so I don't know if TDM400P will work or not.<br>
<br>
Here is the flow when I am using X100P to dial out.<br>
1. Pick up phone<br>
I hear tone. DA~~~<br>
2. press the number <br>
tone: DA~~~<br>
3. dialing~~~~<br>
No more tone. Music playing~~~~~(lalala, I love lalal)<br>
At the same time, on asterisk console, it prints out. "The call has
been answered". <br>
Actually it is still dialing and my mobile is ringing because I didn't
answer the call.. The music was played by ISP <br>
4. whether I answered the call or refuse the call. No more prints on
asterisk console. <br>
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.".<br>
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. <br>
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.<br>
<br>
So can TDM400X work with such a system without tone only with music and
voice?<br>
<br>
Thanks.<br>
Regards.<br>
Scott<br>
</div>
</div>
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