Newbie Sip Trunking question


#1

I want to implement something where people would call a number, and based on the number called, bridge that call to another phone number and record the call.

The customer’s phone system can not be altered, so I was hoping to use a sip trunking provider to provide the incoming and outgoing calls from the PBX.

That being said, how many trunk lines would I need to bridge 50 simultaneous calls… 100?


#2

In SIP, there’s not a physical tying of one trunk to one simultaneous call, as there would be with analog telephony (one circuit to one call) or digital telephony (one bearer channel per call). If you want 50 simultaneous calls, you tell your provider you want to do 50 simultaneous calls.

I might not be fully understanding your proposal, but if you are indeed tying into a traditional PBX and wanting to push it out to SIP, and that traditional PBX does not already support SIP (you’re using Asterisk to get there), then you’ll need some kind of connection from the traditional PBX to Asterisk that can support 50 or 100 simultaneous connections. In this case, you’re going to be pushing E1s (most of the world, 30 channels / simultaneous calls per E1) or T1s (North America, 23 or 24 channels / simultaneous calls per T1) from that traditional PBX to Asterisk. For that, I, naturally, recommend Digium’s multi-port digital E1/T1 interface cards (store.digium.com/telephony_card_selector.php)


#3

You will need one physical trunk line with a capacity of around 10Mb/s (allowing some margin).

You will need a true DID with however many different incoming numbers you have.

As long as the service provider was providing a service for PABX, rather than domestic use, the only reason the service provider might care is for capacity planning, to ensure they didn’t overload any of their equipment. The more simultaneous calls you make, the bigger their revenue!

This should have been asked on Asterisk Support.