[Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Stanislav Sinyagin
ssinyagin at yahoo.com
Mon Sep 9 01:44:55 MSD 2013
Markus, how "the PSTN" would know the length of the destination (German) phone number?
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.
just a few examples:
http://www.de-cix.net/contact/contact/
https://www1.pbxes.com/iptel_impressum.html
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?
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.
________________________________
From: Markus Müller <freeswitchdev at priv.de>
To: freeswitch-dev at lists.freeswitch.org
Sent: Sunday, September 8, 2013 2:41 AM
Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
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.
>
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>________________________________
> 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>
>>To: 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
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
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>>http://www.freeswitchsolutions.com
>>
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>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>FreeSWITCH-powered IP PBX: The CudaTel Communication
Server
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>Official FreeSWITCH Sites
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>_________________________________________________________________________
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