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

Michael Jerris mike at jerris.com
Mon Sep 9 18:49:26 MSD 2013


You can configure this all in freeswitch dialplan if freeswitch is answering, simply don't answer first, you can return incomplete, you can just bridge to something else that will return incomplete and that will pass across the bridge.

Mike


On Sep 9, 2013, at 10:31 AM, Markus Müller <freeswitchdev at priv.de> wrote:

> Hi Mike,
> 
> thanks for helping against the unteachable ignoramus here... I am very bugged about this SHIT!
> 
> 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?
> 
> 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?
> 
> Regards,
> Markus Mueller
>> 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 at 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 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:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
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>>>>> 
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>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
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>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
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>>>>> 
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>>>> 
>>>> 
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
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>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
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>>> 
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>>> 
>>> 
>>> 
>>> 
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>> 
>> 
>> 
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
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>> 
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> 
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
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