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.
Mike
On Sep 7, 2013, at 8:41 PM, Markus Müller <freeswitchdev@priv.de> wrote:
_________________________________________________________________________Hi Stanislav Sinyagin,
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.
====>> 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.
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.
Regards,
Markus Mueller
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.
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.
From: Markus Müller <freeswitchdev@priv.de>
To: freeswitch-dev@lists.freeswitch.org
Sent: Saturday, September 7, 2013 10:08 PM
Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
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.hi Markus,
how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that?
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.
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?
No comments about my design suggestion?
Regards,
Markus
From: Markus Müller <freeswitchdev@priv.de>
To: freeswitch-dev@lists.freeswitch.org
Sent: Saturday, September 7, 2013 1:52 PM
Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides
with the design of a dialplan: it sends the numbers the user types into
his phone LIVE (!) through the ISDN network. If the number is complete,
the ISDN network tells it to the caller; Only now the call get
established (means the dialplan gets invoked). So you have to do an own
step (live and interactive determination of the number) BEFORE the
dialplan comes in line.
Because this is object not supported by freeswith, in the following
situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <->
Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds
until the user has entered the last number and then it goes directly to
the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means,
freeswitch should relay each number the user types into his phone to the
ISDN on the remote side, and make the dialplan stuff AFTER the number
has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I
would code this the following way into FreeTDM: If I get a call with a
starting number I know that it must go to the external ISDN (means: a
second dialplan), I first relay the typed numbers and determine the full
number. Only now, when I got the full number, I would give this call to
the higher layers.
What you think about this?
Regards,
Markus Mueller
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org