[Asterisk-biz] Conference Bridge Application and Incoming Calls

Jonathan Weinberg baselineman at hotmail.com
Sun Nov 21 08:22:39 MST 2004


Hi all, I was wondering if I could ask for some help.  I am new to Asterisk
and have spent time reading the wiki, searching archives and lurking around,
but the time has come for me to come out of the closet so-to-speak and seek
out some answers.  Apologies if this has been adressed before on the list.
 
Basically I am looking to strart up a conference bridge service.  I am
looking to offer the same type of features that the biggies like AT&T offer
including reservation and reservationless conferences, conference recording,
web control, dial-out capability, etc.  I have a basic installation of
Asterisk installed and have been playing around with the meetme
functionality.  It's pretty good, but would need to be extended in order to
meet many of my needs.  
 
I'm trying to figure out whether to go the buy vs. build route right now.  I
know there is one vendor, Indosoft (http://www.indosoft.com) that makes an
add-on to * for more full featured conferencing functionality.  I was
wondering if there are any others out there who do the same?  I was also
curious if there are any other commercial service-provider type conferencing
bridge services out there that people know of on *?
 
Secondly, once I get my ducks in a row with regard to s/w, I need to make a
decision on incoming service.  I figure the options are to go with quad
T-1's to one or more 1U boxes or to do SIP termination and have no hard
telco connections at all.  I know the SIP route would be less expensive, but
I'm worried about quality.  I would be hosting these boxes in a tier-1
facility with multiple ISP connections on a fast backbone so I know my end
would be pretty good, but I was curious if anyone had any thoughts on this
from a performance/quality standpoint.  Am I being unrealistic to think that
I can launch a commercial service targeted to businesses using only SIP?
 
Lastly, on the subject of SIP termination, I am looking to get ballpark
pricing for incoming 8XX service using SIP termination.  I guess I would
need one or more 8XX DID's, but I'm not sure how it works with regard to
incoming capacity.  Obviously with T-1's, you can fill up the number of 64K
channels you have available, but I'm thinking it's more bandwidth oriented
with SIP?  Do I get a certain number of incoming connections for a certain
cost or does it not matter since with toll-free service it's based on
pricing per minute so the number of inbound calls in unlimited?  Also, any
recommendations on providers for this type of service?
 
Thanks in advance for any help advice you guys can give me!
 
Jonathan
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