[Leaplist] Meeting 3/19/09: Asterisk and VOIP

[Tom Parker]

2009/3/18 John Simpson <jms1 at jms1.net>:

> ...
> (1) what companies out there can sell minutes and/or phone numbers which can
> be assigned to an asterisk box via IP? what's their pricing like and what
> kinds of results have people typically had with them? what web sites,
> forums, wikis, etc. exist for people to compare the providers, or to help
> each other with asterisk-related issues?

A lot of the old CLEC's are starting to support SIP trunks.  Locally,
I know Bright House Business and PAETEC now offer commercial SIP
lines.  I've never used them for that.  Most of my setups have been
with taking a T1 or POTS line and plugging it into a Digium card.  For
smaller setups, I've used these SIP providers with varying degrees of
success:
  * http://broadvoice.com/
  * https://www.teliax.com/ - they can provide IAX trunks, which work
much better through NAT firewalls
  * http://www.voicepulse.com/

I've found this site to be quite handy for Asterisk info:
http://voip-info.org/wiki/view/Asterisk
Most of my Google searches on Asterisk information ultimately lead me there.

> (2) if somebody has one or more existing phone lines, whether via POTS or
> PRI, what are their options if they want to plug those lines into an
> asterisk server instead of a $25,000 nortel system? where can they buy the
> cards needed in order to physically interface the lines to their system? and
> what kind of real-world performance can they expect from a typical asterisk
> server- for example, if somebody has a PRI with 8 (or 23) voice channels,
> how "big" a machine would they need in order to reasonably be able to handle
> that many calls at once? or maybe with just two to six lines, plus multiple
> voice terminals (i.e. phones on peoples' desks) carrying "intercom"
> conversations?

Digium or Sangoma are your best bet.  If you can, try to stick with
Digum.  I only say that because that is the company that does the bulk
of the Asterisk development and I'd like to see them stay around.
Sangoma does have some more unique products, so they may have a niche
that Digium cannot fill yet.  I've only seen their products at the
Astricon trade show and I've had no other experience with them.

I personally like to buy my VoIP hardware from The VoIP Connection.
They are over in Melbourne, so shipping is cheap and quick.
http://www.thevoipconnection.com/store/catalog/Digium-Cards-orderby0-p-1-c-265.html

Intercom features on VoIP phones can be tricky.  You have to configure
SIP headers, plus the receiving phone has to support it.  Each
manufacturer seems to have their own opinion on how to implement it.
I've been able to make it work with Snom and Polycom phones.  You
basically have to set the phone to auto-answer when a certain SIP
header is received.

Sizing an Asterisk PBX is pretty easy.  The hardware requirements are
minimal.  What you need to think about is how many channels are going
to be transcoded.  If you have a call coming off of a Digum card using
the G.711(ulaw) codec and you are sending it to a phone with no other
transcoding, you will find there is very little CPU demand.  If that
call needs to be converted to G.729, which cuts back on bandwidth
usage, you need to make sure you have some processing power.  I have
an office of 30 phones and meet-me conferencing running on a Core 2
Duo processor and the CPU utilization has never been an issue.

> (3) i'm pretty sure you can set up a network of multiple asterisk servers
> which are linked together somehow- for example, one in orlando, one in
> tampa, one in jacksonville, and so forth. many companies have a "dial 9 for
> an outside line" policy, is it possible to set up something like "dial 81
> for an outside ORLANDO line", "dial 82 for an outside TAMPA line", and "dial
> 83 for an outside JACKSONVILLE line"? and can you set up something where
> somebody dialing into the local orlando number can dial a code, and then be
> patched through to make an outbound local call from the jacksonville number?
> or even better, where the outbound line is selected based on the number
> you're calling- for example, if you're dialing a 904-786 number (jax), it
> would use a jacksonville line to make the call, thereby saving toll charges?

That's completely possible with Asterisk, but there is a more elegant
way to do this.  If you want to automate it, look into a protocol
called DUNDI (http://dundiglobal.org/).  This allows you to share dial
plans with other servers and find the best way to reach a phone
number.

> i personally want to build a machine which handles one POTS line, plus one
> VoIP line, and set up a way so that when calls arrive on the POTS line, if
> they're from certain numbers they are automatically forwarded to my cell
> phone, otherwise they get a menu... and where i can call from my cell phone,
> dial a code, and call out through the POTS line. (the "POTS" line here is
> actually a vonage adapter with a british phone number, this will enable my
> client in the UK to reach my cell phone without dialing an international
> call, and allow me to call them from my cell phone.)

You would need one of these:
http://www.thevoipconnection.com/store/catalog/Digium-Wildcard-TDM410P-FXO-FXS-Interface-Card-p-16185.html
... plus one of the FXO modules.

FXO modules connect to the phone company, FXS modules connect to the
phone.  That molex plug on the card is only needed if you are
providing analog phone service with an FXS module.

> and of course i have several clients who are interested in knowing more
> about my being able to set them up with an asterisk server, so long as i
> know how to make it work and i'll be able to support it.

In my opinion, a phone system needs to have the ability to allow end
users to do certain things on it.  Like the ability to check your
voicemail via web browser, make extension changes, allow/disallow
certain types of calls, view call detail logs, etc.  Building an
Asterisk system from the ground up is not a simple task, especially if
they want something like conference calling or a complex IVR system.
Because of this, I ended up using a product called Switchvox.

Switchvox (http://www.switchvox.com/) is a Digium company as of about
a year ago and it really takes away a lot of the headaches in the
setup.  The big drawback is that it cuts you off from the flexibility
you have from being able to edit the Asterisk configuration files by
hand.  I quickly got over that when I found out I could still
integrate it with another Asterisk server and do all of my
customizations there, plus they have a really cool HTTP-API for tying
it in with a database.

Using the HTTP-API, you could do some nifty things like having a user
call in to check the status of the invoice they submitted.  Switchvox
will get the invoice number via DTMF and then submit it to the
database server via a URL like
http://acctserver/invoices?apikey=abcdefabcdef&invoice_number=123456.
The database server will reply with the status and the Switchvox
server can then speek the status based on the results.  I never set
this up in a production environment, but I have gotten it to work
using Ruby on Rails.

John, if you have some serious interest in this for some clients, I
would be more than interested in teaming up with you on a project.
Contact me off-list if you are interested.

Tom

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