asterisk users - Oct 2003 - Re: Large installation [was: SS7 signalling/Softswitch]

>I spoke with someone today who is interested in an IP Centrex solution that
>starts with about 3500 extensions in a multi-tenant application.  And
>growing from there.
>
>I'm wondering about scalability of Asterisk.  I'm trying to put my
head
>around how to put the whole thing together, if it can be put together.
>
>The nice thing about it is that if I can show potential, functionality, and
>scalability, which is something I'm starting to see (a recent
contributor
>indicated 240 simultaneous calls), the deal will mean more development
>dollars for adding fine features to Asterisk.  If I play my cards right, we
>might be able to get the engineering info from Cisco we need to make the
>Skinny phones work in all their true, cool functionality.
>
>And to continue with SS7 conversations, I think this gets a good tie in for
>SS7 for handling numerous distributed gateways and Telco interactions.
>
>So a number of questions:
>1) so far, I've heard 240 simultaneous calls.  Does anyone have systems
that
>are larger?
>2) does anyone have suggestions on where to go for making SS7 / Asterisk
>integration a reality?  Obviously on a paid basis.
>3) can what I'm proposing work, or am I off my rocker?
>
>Obviously there are a bunch of things like redundancy, load balancing, load
>management, etc that need to be engineered, but I just wanted to be sure
I'm
>going in the right path.
>
>For instance, Jeremy, do you have statistics you'd like to publicize in
>terms of the number of callers you have, number of active extensions in you
>extensions.conf file, number of minutes/channels/... you put through your
>system?  How much of it is Asterisk based and how much is simply gateway
>calls?
>
>Regards,
>Ray Burkholder
>www.oneunified.net
>704 576 5101
Ray -
   This is a significant change from the topic of softswitches, so I 
re-titled and started a new thread, even though the letters "SS7" do 
appear in your notes.

   While the description of Asterisk as an "Open Source PBX" is 
somewhat descriptive, the system can do significantly more than the 
typical PBX.  However, it is not an "Open Source Softswitch" yet, and 
so SS7 is not an option now nor do I expect it to be in the near- to 
mid-term future without some miracle occurring.  Asterisk can make 
route decisions based on what interface a call might take (SIP, H323, 
PRI, analog, etc.) in as sophisticated a way as you're able to hack 
up in perl/python/whatever.  Routing across multiple systems to build 
a very large PBX is certainly possible, especially with IAX.

  To answer your questions specifically:

1) 240 calls of what kind?  Internal or external?  Using what?  How 
many gateways to the PTSN do you have, and where are they?  VoIP-only 
systems that are just packet forwarders (no transcoding) I'm sure can 
handle >240 calls.  I strongly suspect that any more than 2 4-port 
PRI cards in a single system is asking for trouble, though.

2) Suggestions for making SS7 integration a reality: get a nice tidy 
$100,000 in an account somewhere and hire programmers to produce 
professional software.  That number is not an exaggeration.  You 
might be able to do it for free; but are you willing to stake the 
life of your company on results of a non-paid group of programmers? 
Open Source is amazing and robust, once the code is written.  If the 
code isn't written, I would not suggest that it could be done for 
free or even cheaply.

3) Yes, what you're proposing can work, with adequate planning and a 
very seasoned network and asterisk jockey (or a very understanding 
client who can wait for functionality.)  What %age would be using 
VoIP?  What %age using analog-to-T1 conversions (channel bank/T1 
interface)?  What %age need their own voicemail, conferencing, etc. 
etc. etc.?  What kind of redundancy/uptime is the client expecting? 
This radically changes the design.

JT