[asterisk-users] SIP vs Analog lines

[David Backeberg]

On Tue, Jul 28, 2009 at 8:18 PM, John F. Ervin<jervin at jervin.com> wrote:
> Never having actually rolled an Asterisk (Trixbox in my case) system into
> production.  I was wondering if in most peoples opinion if given the choice
> would rather have a straight VOIP/SIP system or would rather have a system
> with normal POTS/analog types lines and something like a digium card?  As
> far as reliability etc.  Thoughts?

Number of simultaneous lines?
Budget?
Number of minutes monthly?
Is your greatest expense going to be the hardware, or the phone charges?
What you're planning to do with it?

I've done a lot of work with SIP-based MeetMe conferences, and let me
tell you that DAHDI-based MeetMe seems to 'just work', compared to all
the tuning we put in to get SIP-based, dahdi_dummy -timed MeetMe
working and not sounding bad.

My company priced out several options and determined that our best
pricing was to minimize the cost of calls by purchasing dedicated TDM
hardware and have telco terminate locally. We run SIP in some cases
inside our VoIP gateway gear.

So in answer to your question, we hedge our bets by doing both. We
think that telco is generally rock solid, and we couldn't find a SIP
provider that could compete with the pricing for the TDM-based
solution, especially once you add in the cost of the bandwidth
overhead for all those calls. We have a handful of the 4-port T/E
cards, an old 1-port T card, and our trusty old TDM410 card.

We're now in the league of a channelized DS3, which we break out into
a lot of T1s with some nice Cisco gear. Google the Catalyst 3845 when
you're big enough to need that many lines.