<html><body><div style="color:#000; background-color:#fff; font-family:arial, helvetica, sans-serif;font-size:10pt">I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.<br><br>So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.<br><br><br><br><div><span><br></span></div><div><br></div> <div style="font-family: arial, helvetica, sans-serif; font-size: 10pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Markus Müller <freeswitchdev@priv.de><br> <b><span style="font-weight: bold;">To:</span></b> freeswitch-dev@lists.freeswitch.org <br>
<b><span style="font-weight: bold;">Sent:</span></b> Saturday, September 7, 2013 10:08 PM<br> <b><span style="font-weight: bold;">Subject:</span></b> Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)<br> </font> </div> <div class="y_msg_container"><br><div id="yiv1287044465">
<div>
<div class="yiv1287044465moz-cite-prefix">Hi all,<br>
</div>
<blockquote type="cite">
<div style="color:#000;background-color:#fff;font-family:arial, helvetica, sans-serif;font-size:10pt;">hi Markus,<br>
how do you determine if it's a complete number? Do you expect
the outgoing ISDN channel to tell you that?<br>
</div>
</blockquote>
Sorry that I don't have all details; I am not as familary as I want
to with the ISDN protocols. But I know that each number goes one by
one to the outgoing ISDN and then outgoint ISDN tells you back when
you have reached a destination. If needed I can make a log when I
have my Siemens Hipath placed behind my Freeswitch in NT mode; there
you can see the numbers comming one by one.<br>
<blockquote type="cite">
<div style="color:#000;background-color:#fff;font-family:arial, helvetica, sans-serif;font-size:10pt;"><br>
I also wonder what happens if you attach an ISDN-SIP gateway,
like Patton. Will you have a new SIP message (which?) on every
dialed digit?<br>
</div>
</blockquote>
Normaly all the products do the "overlap dialing" crap, as mentioned
in my first email. I don't know for sure but I think there is no
alternative, especially if you make only SIP and not ISDN.<br>
<br>
No comments about my design suggestion?<br>
<br>
Regards,<br>
Markus<br>
<blockquote type="cite">
<div style="color:#000;background-color:#fff;font-family:arial, helvetica, sans-serif;font-size:10pt;">
<div><br>
</div>
<div style="font-family:arial, helvetica, sans-serif;font-size:10pt;">
<div style="font-family:times new roman, new york, times, serif;font-size:12pt;">
<div dir="ltr">
<hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Markus
Müller <a rel="nofollow" class="yiv1287044465moz-txt-link-rfc2396E" ymailto="mailto:freeswitchdev@priv.de" target="_blank" href="mailto:freeswitchdev@priv.de"><freeswitchdev@priv.de></a><br>
<b><span style="font-weight:bold;">To:</span></b> <a rel="nofollow" class="yiv1287044465moz-txt-link-abbreviated" ymailto="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>
<br>
<b><span style="font-weight:bold;">Sent:</span></b>
Saturday, September 7, 2013 1:52 PM<br>
<b><span style="font-weight:bold;">Subject:</span></b>
[Freeswitch-dev] Call relaying (ISDN <> Freeswitch
<> ISDN)<br>
</font> </div>
<div class="yiv1287044465y_msg_container"><br>
Hello Freeswitch Developers,<br>
<br>
ISDN in germany (maybe also somewhere else) has a feature,
which colides<br>
with the design of a dialplan: it sends the numbers the
user types into<br>
his phone LIVE (!) through the ISDN network. If the number
is complete,<br>
the ISDN network tells it to the caller; Only now the call
get<br>
established (means the dialplan gets invoked). So you have
to do an own<br>
step (live and interactive determination of the number)
BEFORE the<br>
dialplan comes in line.<br>
<br>
Because this is object not supported by freeswith, in the
following<br>
situation<br>
<br>
User <-> Analog Phone <-> Siemens Hipath
<-> ISDN <-> [FreeTDM <-><br>
Freeswitch <-> FreeTDM] <-> ISDN <->
World <-> ISDN Destiation<br>
<br>
you have to do "overlap dialing". Means, freeswitch waits
some seconds<br>
until the user has entered the last number and then it
goes directly to<br>
the dialplan.<br>
<br>
-> This is not what I need!<br>
<br>
I want that it works as if there is no FreeSwitch in
between. Means,<br>
freeswitch should relay each number the user types into
his phone to the<br>
ISDN on the remote side, and make the dialplan stuff AFTER
the number<br>
has been dicovered for completeness.<br>
<br>
How do you think this should be implemented? If nobody has
an object, I<br>
would code this the following way into FreeTDM: If I get a
call with a<br>
starting number I know that it must go to the external
ISDN (means: a<br>
second dialplan), I first relay the typed numbers and
determine the full<br>
number. Only now, when I got the full number, I would give
this call to<br>
the higher layers.<br>
<br>
What you think about this?<br>
<br>
Regards,<br>
Markus Mueller<br>
<br>
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