[Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)

[Markus Müller]

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 at priv.de>
> *To:* freeswitch-dev at lists.freeswitch.org
> *Sent:* Saturday, September 7, 2013 10:08 PM
> *Subject:* Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
>
> Hi all,
>> hi Markus,
>> how do you determine if it's a complete number? Do you expect the
>> outgoing ISDN channel to tell you that?
> 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.
>>
>> 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?
> 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.
>
> No comments about my design suggestion?
>
> Regards,
> Markus
>>
>> ------------------------------------------------------------------------
>> *From:* Markus Müller <freeswitchdev at priv.de>
>> <mailto:freeswitchdev at priv.de>
>> *To:* freeswitch-dev at lists.freeswitch.org
>> <mailto:freeswitch-dev at 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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>>  </>
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>
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>
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