<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">What your searching for IS overlap dialing. And it works by sending the setup and receiving the error code for incomplete. This should pass across a bridge, you can also handle this yourself in the dialplan if you like. If this is not passing across a tdm to tdm bridge, that is a bug and should get filed in jira.<div><br></div><div>Mike</div><div><br><div><div>On Sep 7, 2013, at 8:41 PM, Markus Müller <<a href="mailto:freeswitchdev@priv.de">freeswitchdev@priv.de</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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<div class="moz-cite-prefix">Hi Stanislav Sinyagin,<br>
<br>
what you mean is the "overlap dialing" feature, this is already
implemented and working in freetdm. It waits some seconds after
the last number and then dials.<br>
<br>
====>> This is not what I am searching for ! This is also
not what I am talking about ! Please accept this and make sure how
the reality is before saying that I am telling wrong things.<br>
<br>
I am searching what I worte, if you believe or not. ISDN in
germany currently works this way, and yes, there is a
"completeness" of numbers which is interactively determined by
sending each key through the ISDN network and getting back when a
destination has been found.<br>
<br>
Regards,<br>
Markus Mueller<br>
<br>
</div>
<blockquote cite="mid:1378595113.21724.YahooMailNeo@web126203.mail.ne1.yahoo.com" type="cite">
<div style="background-color: rgb(255, 255, 255); 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 size="2" face="Arial"> <b><span style="font-weight:bold;">From:</span></b> Markus
Müller <a class="moz-txt-link-rfc2396E" href="mailto:freeswitchdev@priv.de"><freeswitchdev@priv.de></a><br>
<b><span style="font-weight: bold;">To:</span></b> <a class="moz-txt-link-abbreviated" 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 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="background-color: rgb(255, 255, 255); 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="background-color: rgb(255, 255, 255); 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">
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<div><br>
</div>
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sans-serif;font-size:10pt;">
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york, times, serif;font-size:12pt;">
<div dir="ltr">
<hr size="1"> <font size="2" face="Arial">
<b><span style="font-weight:bold;">From:</span></b>
Markus Müller <a moz-do-not-send="true" 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 moz-do-not-send="true" 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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<pre wrap="">_________________________________________________________________________
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_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-dev mailing list<br>FreeSWITCH-dev@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>