Hello, I'm trying to configure our Inter-tel AXXESS (R4.4) system to connect
to an Asterisk system (0.7.2 for now) to initially be a conference bridge,
and perhaps later start hanging VoIP phones off of it so that we can
gradually phase out the AXXESS system (we don't have the budget to do a
forklift upgrade).
I've added an extra T1C (non-PRI) card to the AXXESS, set up a PC with a
Digium T100P, and connected them with a crossover T1 cable. That part works.
If I configure the trunk lines on the AXXESS to Loop Start and Asterisk to
fxo_ks, I get a dialtone from Asterisk when I dial the trunk group from an
AXXESS keyset, and from there I can dial an extension on the Asterisk system.
It also works if I use E&M signalling (I get the demo application).
Here's a cheesy diagram of how it's laid out:
______
( PSTN ) ___________ ______________
(______)===E&M T1===| Inter-Tel |===T1===| Asterisk |
| AXXESS | | Meetme x1300 |
|___________| |______________|
|
Axxess keysets:
x1000 - x12XX
I'd like to get 4-digit dialing working between the two systems, so that
users
can dial x1300 on the AXXESS and connect to the MeetMe app. I figure I can use
*ANI*DNIS* for that. I'm pretty sure I can get the AXXESS to receive
*ANI*DNIS* (at least using E&M signalling - we receive DNIS on the T1 from
the
PSTN already), but I'm not so confident that it'll send it. I'm
asking
Inter-Tel about that part (as well as ARS, that I apparently need to configure
to get it to send the call down the trunk without dialing the trunk group
manually in the first place). Asterisk has to be able to send and receive
*ANI*DNIS* though, and I haven't been able to find any documentation as to
how
that's done. In my naivety I've tried stuff like:
exten => 1005,1,Dial(Zap/g1/*9204*1005*)
It places the call, but the 'E' pattern on the AXXESS call routing table
gets
matched, meaning no digits were received.
Without *ANI*DNIS*, I guess I could pass calls around using DISA, but then
people wouldn't know what extension is calling them. Maybe I can live with
that just for conference bridging, but if I start phasing in VoIP phones,
people will want to know who they're talking to (they're used to it).
Also we have a limited number of trunk lines to the PSTN and we don't want
all
of them to be used for conferencing. Several of the people in the conference
may be connecting from the AXXESS system instead of the PSTN so it'd be
suboptimal if we had to limit it by limiting the number of trunk lines to the
Asterisk system. Obviously if I can't get inbound ANI info, Asterisk
can't
differentiate between calls from the PSTN and local calls.
So, my questions are:
* Anyone done this before?
* Is *ANI*DNIS* a workable option, or will 4-digit dialing not work without
PRI?
* Will it work with PRI? :)
* Is it more workable to connect the AXXESS system to the PSTN by way of the
Asterisk system?