[asterisk-users] Asterisk to Avaya

Tom Lynn tom at tomlynn.com
Tue Aug 5 19:41:42 CDT 2008


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
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080805/a5ec0c1d/attachment.htm 


More information about the asterisk-users mailing list