<html><body><div style="color:#000; background-color:#fff; font-family:arial, helvetica, sans-serif;font-size:10pt">Markus, how "the PSTN" would know the length of the destination (German) phone number? <br>I think it's just impossible, especially that there are multiple independent providers in Germany, and the phone numbers differ in length even within the same end customers.<br><br>just a few examples:<br>http://www.de-cix.net/contact/contact/<br>https://www1.pbxes.com/iptel_impressum.html<br><br><div><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;">So, I still think somewhere along the chain there's a simple interdigit timeout. Probably it's in your outgoing ISDN channel at the provider side. How else would the system know when to place the call?</div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family:
arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;">So, instead of relying on someone else's interdigit timeout, you can handle your own, and place the outbound call when you have enough digits.</div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><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> Sunday, September 8, 2013 2:41 AM<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="yiv6872311282">
<div>
<div class="yiv6872311282moz-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 type="cite">
<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 <a rel="nofollow" class="yiv6872311282moz-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="yiv6872311282moz-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 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="yiv6872311282y_msg_container"><br>
<div id="yiv6872311282">
<div>
<div class="yiv6872311282moz-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="yiv6872311282moz-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="yiv6872311282moz-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="yiv6872311282y_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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