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Questions and Answers from Asterisk Presentation

>i'd like a basic rundown on setup for a home office with one "real"
>external phone number / line that routes if possible to a number of
>internal "virtual numbers / lines / vboxes".
>
>in that scenario, are the internal virtual lines / boxes only accessible
>from a computer with mic & headset or can they be routed to actual physical
>"internal" phones with "local dummy numbers" behind the asterisk pbx?  any
>hints on related configuration would be helpful.

hard versus soft phones

>can windows clients access the vboxes on the asterisk server?

Web interface to voicemail

>based upon your experience, any notes on gotchas related to configuration
>and regular operation of the system?

>how does one handle a failure on the asterisk server?  at a bare minimum,
>i'm assuming you unplug the server and plug in a physical phone into the
>wall jack to get a single working line, foregoing all of the extra virtual
>lines and vboxes until you can bring the asterisk server back up.

That would work for a hard POTS-mostly system.  For VoIP/SIP you may have
troubles.

You can set up HA...

=================

>That said:  There are applications it does NOT do well.  For example, it
>does conference bridging, but not to professional quality standards. The
>latency it introduces into the audio stream doesn't meet any carrier's
>specifications for quality.  (Most carriers want audio energy transported

=================

>It would be interesting (to me) to know, at a high level, what options
>are feasible for tying into VoIP systems and services. I see a newly
>announced "Skype for Asterisk" product as well as support for SIP and
>other protocols. Is this generally mature and if someone at home or a
>small office were looking for voip on the cheap, what's the best way
>to leverage Asterisk to provide this?

VoIP on the cheap doesn't need Asterisk.
But, there are providers that will talk with Asterisk or SIP.
Asterisk just makes it better, at the cost of more complexity.

=================

>Does the home need to be star-wired, punch downs and all, to benefit from
>Asterisk?

No.  I did, but now I'm just doing SIP hard phones over IP.

>I like the question another reader asked related to call filtering - Can
>all calls from a given list of numbers make it through automatically, the
>rest get routed to automated message capture?

Yes.

>Family message mailboxes and custom greetings?

Yes.

>Remote management?

Yes

>Integration with any home automation systems?  (I want to be able to call
>in, turn on my electric blanket...that sort of stuff.)

You can call scripts based on phone input.  Festival can give text to
speech.

=================

>a) does Asterisk work ok on the internet or does it require full
>control of the network to configure all the proxies, firewall, etc
>placed by (at least) the two ISP involved? Does it use some of the
>"tricks" skype uses?

We run it over the Internet, but realize that if your line gets saturated
your phone stops working.  QoS *MAY* help, but usually you only have
control of one side of the line for implementing that.

>b) I've seen that there is some video support for Asterisk, but from
>what I read the video support seems cripple. Do you have experience
>with it?

Nope.

>c) conferencing (and video-conferencing) as already mentioned by others.

Conferencing works ok.  No idea about video.

=================

>Personally, I'd like to know if I could use Asterisk as a filtering
>system on our land lines at home.  I'd like to be able to either disable
>the phones at certain times or pick up the phone when it recognizes the
>incoming number and play an "out of service" tone.  This is to try and
>prevent a collection agency from annoying us (they've got the wrong guy
>but haven't taken us off their automated calling system).

exten => s,n,GotoIf($["${CALLERID(num)}" = "9705551234"]?ToPhone)
exten => 1,n,GotoIfTime(9:00-18:00|mon-fri|*|*?ToPhone)
exten => 1,n(ToVoicemail),Voicemail(s1)
exten => 1,n,Hangup()
exten => 1,n(ToPhone),Dial(SIP/desk-1,25,t)
exten => s,n,Goto(call-${DIALSTATUS})
exten => s,n(call-NOANSWER), Goto(ToVoicemail)
exten => s,n(call-BUSY), Goto(ToVoicemail)
exten => s,n(call-), Goto(s,call-NOANSWER)
exten => s,n, Hangup

>Also, what kind of off the shelf PC hardware works with this?  The
>Asterisk site lists Digium's harware but I was wondering if simply modem
>cards would suffice for the simplest configurations.

You can't just throw a random modem card at it and expect it to work.
Try a TX100/TX101 on ebay, around $30.  I quickly moved to Digium hardware.

=================

>- analogue adapters FXS/FXO
>  - good/bad ones
>  - where to buy

The TX100/TX101 on ebay is a good place to start, but it really depends on
the exact card.  Spending $30 rather than $10 may be worth it if they seem
to really support Asterisk.

>- use with existing voip providers like Vonage, etc. whats new with Skype

I don't know about Vonage specifically, check the voip-info.org forums for
information about specific providers.  Providers that support IAX probably
will work, SIP providers MAY work but may not provide support.

>- implementing your own voip
>  - inbound gateway providers
>  - outbound options

No problem.

>- voicemail, what do you really need to make it work

Very little.  Pretty much works out of the box.

>- security issues
>  - configuring your router...
>  - gotchas

I just run everything over OpenVPN to secure it.  Have set up OpenWRT boxes
to nail up an OpenVPN connection to our Asterisk server and plug phones
into them.

>- voip performance, bandwidth vs jitter
>  - qos/mta issues 

You probably should implement QoS, particularly if you saturate your line
much.  Our Asterisk server is on a gigabit network, so no issue there.  But
your home connection would need QoS on both sides of the line.

I have had few performance issues.

=================

>I'd be interested to know how this interfaces to the telco network.  What
>hardware is required if you have a PSTN line, T1, ISDN, Stand alone DSL,
>fiber?

You need an appropriate adapter.  DSL is going to be an IP connection, so
if you want to access the PSTN over it you need to have a VoIP provider.

>How does one go about obtaining a phone number to use with such a
>device?

Use your existing POTS or use a VoIP provider.

>Once I have "service" to the box - what kinds of phones / devices can I use
>with it?  Standard alalog phones (like cheapy target ones)?  VoIP phones
>(Cisco, Polycom)?  Softphones? etc.

Yes, any of those, provided you have appropriate service (FXS/ATA for cheapy
analog phones, SIP over IP for VoIP hard phones and Soft phones...

>How does this all work with a "real" firewall or NAT device?
>what ports are used?  etc etc...

NATing can cause problems, particularly with SIP.  Bumping up the UDP
packet timeout seems to solve most of those problems.  I run over a
non-NATed OpenVPN to get good security so I don't have these issues.

For ports, see the Wikipedia page on the specific protocol you want to use.