A service provider's configuration of Asterisk

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

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