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<div class="moz-cite-prefix">Hi Mike,<br>
<br>
thanks for helping against the unteachable ignoramus here... I am
very bugged about this SHIT!<br>
<br>
Currently in default this works as you described; I type the "0"
of "089xxxxxx". This goes to the SIP gateway (=Freeswitch), it
send it out on the external ISDN, and this answeres something. It
might be "incomplete"... don't know. In any case: The calling is
aborted by the SIP gateway (=Freeswitch) and there is currently
only the possibility to enable this crap "overlap dialing" feature
in freeswitch (it just waits some seconds if another digit comes
and then dials after the last). If it is just wrong implemented:
Fine! Lets fix it. But... Don't you think it makes it necessary to
change the dialplan behaviour at all? Currently it is fully based
on complete numbers; it just cannot handle incomplete numbers, is
it?<br>
<br>
In any case: I'll do an bug report with log files and so on. BUT
PLEASE: I cannot find anything anymore about creating pcap file.
Does anybody have a hint how to do this currently with
FreeTDM/libpri?<br>
<br>
Regards,<br>
Markus Mueller<br>
</div>
<blockquote
cite="mid:29FD1E69-2C42-4F4E-90B5-8D1E1628CF27@jerris.com"
type="cite">
<meta http-equiv="Content-Type" content="text/html;
charset=ISO-8859-1">
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
moz-do-not-send="true" href="mailto:freeswitchdev@priv.de">freeswitchdev@priv.de</a>>
wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
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http-equiv="Content-Type">
<div text="#000000" bgcolor="#FFFFFF">
<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 moz-do-not-send="true"
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 moz-do-not-send="true"
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">
<div style="background-color: rgb(255,
255, 255); 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 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>_________________________________________________________________________
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<pre wrap="">_________________________________________________________________________
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