On Mon, January 9, 2006 14:03, Daniele Orlandi said:
> On Monday 9 January 2006 11:08, you wrote:
>>
>> I partially understand, but for me this will work, so I'll keep the
>> patch
>> for myself and apply whenever updating to a newer patch level! ;-)
>
> I may be adding a workaround based on a filter which specifies which
> extensions do support overlap dialing so, it won't last for too much. But
> I
> have to work on the EC, first :)
Maybe you could make it an entry in the visdn.conf, based on the same
format as the dial rules... I'll think on it and see whether I can figure
out something helpful... ;-)
>
>> (Partially because I do not 'quite' grasp what you mean by "The problem
>> is
>> that when the overlap dialing fall in an extension which does support
>> overlap dialing I have to start the PBX immediately and pass further
>> digits as DTMF frames")
>
> Suppose you have a box with two ISDN interfaces configured for
> passthrough.
> One in TE mode connected to the telco, one in NT mode connected to an ISDN
> phone.
>
> When you hang off the phone it sends an empty SETUP, asterisk starts the
> PBX
> on the s,1 extensions which in turn has VISDNOverlapDial() application
> that
> also provide the dialtone.
>
> Then I start dialing the digits one by one.
>
> Suppose you have an entry like this in your dialplan and you want to dial
> 01234:
>
> exten => _0.,1,Dial(VISDN/visdn.telco/${EXTEN})
>
> After you digit '0', I have to start Dial() which will send a SETUP with
> '0'
> towards visdn.telco interface and start overlap dialing on it.
>
> All further digits (coming as INFORMATION messages) will be passed between
> channels as DTMF frames and recoverted to INFORMATION messages again.
>
> When the dialed channel receives a PROCEEDING, I pass a PROCEEDING frame
> back
> and a PROCEEDING message on the dialing channel.
>
> This works well and makes ISDN-ISDN calls really transparent.
>
> Now suppose you have another extension and the only valid SIP address is
> 2987:
>
> exten => _2.,1,Dial(SIP/${EXTEN})
>
> You start dialing 2 and Dial(SIP/2) is run, which does not correspond to
> any
> valid address and Dial exit with NOT FOUND...
Now I get it, but this scenario failed on my asterisk box as well...
(It dialed VISDN/visdn0/0, and then nothing until congestion from the PSTN)
But it is possible that that is [EMAIL PROTECTED] / AMP related...
Will investigate that for your benefit!
>
>> Another thing I am looking at right now is that the chan_visdn doesn't
>> recognise the DTMF after the call is established, which means I cannot
>> use
>> functions like ATXFER, XFER, AUTOMON, etc., because these rely on
>> key-presses (resp. *2, # and *1 by default), or even terminate the echo
>> feedback test (*43) by pressing #...
>
> Yes, this is known, I just have to enable the DTMF detecton integrated
> into *
> and maybe use the hardware DTMF detector in some hardware.
>
> There is a TODO entry already, I'll work on it just after the EC.
Cool!
I'll contribute wherever I can, if only by sharing the info I learn...
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
AMD Duron 1GHz - 1GB - * 1.2.1
2 Sweex HFC-PCI cards
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