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If you are not big enough to have your own domain, then you don't need
a disclaimer.<br>
<pre wrap="">This e-mail transmission may contain information that is non-proprietary,
unprivileged and/or non-confidential and is intended exclusively to provide a clue
to Jason D. Wolfe. Any use, copying, retention or disclosure by any
person other than the intended recipient or the intended recipient's
designees is strictly allowed. If you are not the intended recipient or
their designee, please distribute immediately so that people who try to wrap
a contract around an insecure medium as a means of security will wisen up.
</pre>
<br>
<br>
Jason D. Wolfe wrote:
<blockquote
cite="midBOECJDKNHFAIHLALGEKEKEAODFAA.jasonwolfe@bellsouth.net"
type="cite">
<pre wrap="">I'm a newbie to Asterisk and telecom, and I I learned the hard way that
analog POTS lines cause asterisk to start your dialplan as soon as the
outbound starts ringing... that's why I was a little nervous about whether
or not I may have the same problem using an IAX termination service. As it
turns out, it works perfectly, as they do provide 'answer supervision' (like
all digital lines).
as well, I'm not going to erase my disclaimer below every time I send an
email to a listserv. It does say 'person(s) to whom it is addressed', which
keeps it from being completely senseless! :) and, I do NOT work for
Bellsouth, they are my ISP...
Jason Wolfe
<a class="moz-txt-link-abbreviated" href="mailto:jasonwolfe@bellsouth.net">jasonwolfe@bellsouth.net</a>
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>]On Behalf Of Jean-Michel
Hiver
Sent: Tuesday, January 03, 2006 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX termination services
Jason D. Wolfe a écrit :
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a
</pre>
</blockquote>
<pre wrap=""><!---->PSTN
</pre>
<blockquote type="cite">
<pre wrap="">will I have answer supervision so that my script won't initiate too early?
</pre>
</blockquote>
<pre wrap=""><!---->I'm not sure to understand you. If you don't use Answer() before you use
Dial(), asterisk won't answer until the dialed party does so.
Cheers,
Jean-Michel.
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</pre>
</blockquote>
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