<div dir="ltr">Steve, what kind of Avaya system is this? They make several.<br><br><div class="gmail_quote">On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies <span dir="ltr"><<a href="mailto:davies147@gmail.com">davies147@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br>
<br>
Sorry this is so long, but I am reasonably desparate.<br>
<br>
I am having real fun with hooking an Avaya system to Asterisk using<br>
ISDN30. I have the ISDN signalling all sorted one way, and can pass<br>
calls from the "real world" (ie. the telco and asterisk) TO the avaya<br>
box, and it accepts that and sets up the call perfectly.<br>
<br>
The problem is that the Avaya box is signalling outbound calls using<br>
an "odd" method, which smacks of an analogue system with ISDN30 bolted<br>
on for a bit of a laugh.<br>
<br>
They send a q931 SETUP message. This contains the correct callerID,<br>
but only the first 1 to 4 of the dialled number's digits - The<br>
remainder of the number is I believe passed through using DTMF!!! From<br>
the look of it they intentionally do not send an IE 161 "Sending<br>
Complete" with the SETUP, so that the far end continues to listen for<br>
these DTMF tones, until it resolves to a legal number.<br>
<br>
My questions for some ISDN expert out there...<br>
<br>
Part 1) I need to receive the number in the SETUP, which I guess will<br>
be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,<br>
and check the dialplan to see if it is a locally terminated number.<br>
Once I am 100% sure it is not local, I can then dial the collected<br>
number through the Telco ISDN channel. Make sense? I think I can<br>
probably handle that. The problem is that I do not know whether I have<br>
received all digits from the Avaya at that point, which leads to...<br>
<br>
Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a<br>
difference) without sending the IE 161 call complete? I thought that<br>
Dial(Zap/G1||D(${INITIAL}))<br>
might send the initial digits using DTMF, and then leave the channel<br>
open so that more DTMF could follow over the now bridged channel. In<br>
fact I get an immediate failure as if the far end thinks I have<br>
finished dialling. Can I assume that libpri does not currently support<br>
this method of dialling? If not, how might it be added? I can hack the<br>
code, I just need suggestions of where to look and how it might sanely<br>
be added :)<br>
<br>
Part 3) It is possible that the Avaya is not using DTMF at-all, and<br>
that it will send more bits of the called-party number using the<br>
D-Channel as you would expect, but Asterisk does not seem to be<br>
waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone<br>
know the Avaya systems well enough to suggest how it might be working?<br>
<br>
Many many thanks for any feedback.<br>
<br>
Regards,<br>
Steve<br>
<br>
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</blockquote></div><br></div>