[Asterisk-Users] A real-life production scenario

[Ryan Tucker]

Since it's all the craze, I might as well post our current Asterisk 
usage.  :-)

EQUIPMENT:
  - Beefyish box (dual Xeon 2.4GHz, gig of RAM, more-than-adequate disk 
space, etc) in a 1U chassis.
  - A second, slightly less beefyish box of specs I don't have handy right 
now, also in a 1U.
  - 2xTE410P

CONNECTIONS:
  - 1 PRI to telco for local outbound/direct-dial inbound, 300 numbers 
attached.
  - 2 PRI to another telco for toll outbound/toll-free inbound
  - 1 E&M T1 to office PBX

We offer VoIP services to our directly-connected customers, ranging from 
simply taking their toll traffic to providing "virtual PBX" services, all 
using Asterisk.  We've done a great variety of things (oddly, all 
customers are not alike)... here's a sampling:

* Connection to our PBX
   Our PBX previously had a T1 in from a telco using an E&M trunk, with 4 
digits on the DNIS.  When we had the Asterisk stuff stabilized, we wanted 
to move over to it ASAP because LD was much cheaper.  (That, and the T1 
wasn't the cheapest T1 we have here...)

   We disconnected one of the extra toll PRI's and, in its place, put the 
T1 from the telco.  We then connected (using a crossover) the PBX to the 
TE410P.  Various switching magic was performed (this was the point where I 
realized it's only getting 4 digits on the DNIS) and inbound calls were 
sent over to the PBX.  Outbound calls from the PBX were switched like our 
VoIP calls.  Following this, we ordered porting of that block of numbers 
over to the inbound PRI.

   The telco did it about 5pm on a Wednesday afternoon with no 
notification.  Unfortunately, I had slightly bungled the exten => entry 
for calls coming in via that route.  Fortunately, it was easy enough to 
fix, and was fixed before I got about the fourth swear word out of my 
mouth.  The CDR file captured the caller ID on the confrangled calls, and 
our support department called them back promptly, and everyone was happy.

* Customer with their own POTS lines wanting VoIP service
   One of our VoIP customers was in the interesting position of wanting the 
phone lines at their office, terminated analogly.  We had a Mediatrix 
gateway in for testing, and decided to deploy it there.  The Mediatrix was 
configured to send inbound calls to the Asterisk box, as well as gate 911 
calls from the Asterisk to the PSTN (so that, when they call 911, it shows 
up with *their* location instead of *ours*).  Calls from the Mediatrix 
successfully make it to Asterisk (with caller ID) where they ring the 
receptionist phone for 10 seconds then go to an 
auto-attendant/voicemail/etc.  The Mediatrix doesn't answer (and therefore 
doesn't pass the call) until around the second ring, which is annoying, 
but them's the breaks.

There's a bunch of other situations as well, but basically, it'll do most 
things.  :-)  -rt

-- 
Ryan Tucker
Network Engineer
NetAccess, Inc.
1159 Pittsford-Victor Road
Bldg. 5, Suite 140
Pittsford, New York 14534
585-419-8200
www.netacc.net