[asterisk-users] Asterisk to Avaya

[Steve Davies]

I am told it is an IP Office 400 series.

I have not been on site physically which does not help.

Regards,
Steve

2008/8/6 Tom Lynn <tom at tomlynn.com>:
> Steve, what kind of Avaya system is this?  They make several.
>
> On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies <davies147 at gmail.com> wrote:
>>
>> Hi,
>>
>> Sorry this is so long, but I am reasonably desparate.
>>
>> I am having real fun with hooking an Avaya system to Asterisk using
>> ISDN30. I have the ISDN signalling all sorted one way, and can pass
>> calls from the "real world" (ie. the telco and asterisk) TO the avaya
>> box, and it accepts that and sets up the call perfectly.
>>
>> The problem is that the Avaya box is signalling outbound calls using
>> an "odd" method, which smacks of an analogue system with ISDN30 bolted
>> on for a bit of a laugh.
>>
>> They send a q931 SETUP message. This contains the correct callerID,
>> but only the first 1 to 4 of the dialled number's digits - The
>> remainder of the number is I believe passed through using DTMF!!! From
>> the look of it they intentionally do not send an IE 161 "Sending
>> Complete" with the SETUP, so that the far end continues to listen for
>> these DTMF tones, until it resolves to a legal number.
>>
>> My questions for some ISDN expert out there...
>>
>> Part 1)  I need to receive the number in the SETUP, which I guess will
>> be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,
>> and check the dialplan to see if it is a locally terminated number.
>> Once I am 100% sure it is not local, I can then dial the collected
>> number through the Telco ISDN channel. Make sense? I think I can
>> probably handle that. The problem is that I do not know whether I have
>> received all digits from the Avaya at that point, which leads to...
>>
>> Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a
>> difference) without sending the IE 161 call complete? I thought that
>>  Dial(Zap/G1||D(${INITIAL}))
>> might send the initial digits using DTMF, and then leave the channel
>> open so that more DTMF could follow over the now bridged channel. In
>> fact I get an immediate failure as if the far end thinks I have
>> finished dialling. Can I assume that libpri does not currently support
>> this method of dialling? If not, how might it be added? I can hack the
>> code, I just need suggestions of where to look and how it might sanely
>> be added :)
>>
>> Part 3) It is possible that the Avaya is not using DTMF at-all, and
>> that it will send more bits of the called-party number using the
>> D-Channel as you would expect, but Asterisk does not seem to be
>> waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone
>> know the Avaya systems well enough to suggest how it might be working?
>>
>> Many many thanks for any feedback.
>>
>> Regards,
>> Steve
>>
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