[GTALUG] VOIP setup

Blaise Alleyne email+libre at blaise.ca
Thu Dec 31 22:50:03 UTC 2015


On 31/12/15 05:42 PM, David Mason wrote:
> I know there are some people on here that know about VOIP and Asterisk, and I'm
> looking for some pointers.  I've looked at Asterisk a bit (a while ago) and it
> was kind of overwhelming.
> 

I'm running a FreeSWITCH server at work. More overwhelming in some ways, less in
others. There's just a lot of telephony stuff you need to get up to speed with...

Though, at home, I'm just using SIP clients, connecting directly to VoIP.ms SIP
servers in Toronto and using their web portal to configure call forwarding,
IVRs, etc.


> I want to have several phones and several phone numbers (a 416 for where I work,
> a 905 for where I live, and a 902 for where a bunch of my friends and family
> are).

VoIP.ms

> I want to pick up a phone, dial a number and have the appropriate VOIP
> phone number be used.

Caller ID is a little trickier, but definitely doable. With SIP and VoIP.ms,
there are two options:

Option 1: Set the Caller ID in your VoIP.ms account profile. This would always
be the same, wouldn't be able to use different numbers for outgoing caller ID.
This is what I do at home.

Option 2: Set VoIP.ms to accept the caller ID from your calls. With my
FreeSWITCH server at the office, we have it set to use different outbound caller
IDs depending on a prefix dialed before the number. Maybe there's a way to set
it in your SIP client without needed an Asterisk/FreeSWITCH server in between
you and VoIP.ms? You'd need to find a way to set this and trigger the
appropriate caller ID on your end though before it hits VoIP.ms, if you want it
to vary.

> I'd like a different ring tone from each of the numbers. 

Also a little tricker, but probably doable. The ring tone is going to be set by
your SIP client, whether it's a softphone or handset.

(a) You'd need a SIP client that can vary the ringtone based on the incoming call.

(b) You'd need a way to identify the number used by the incoming caller.

So, if John Doe calls you on your 902 number, your SIP client won't know that it
was on the 902 number -- it will just see John Doe's name and number.

However, VoIP.ms has an option for DIDs (Direct Inward Dialing numbers, i.e.
traditional phone numbers) to prefix the caller ID. So, at the office, we've got
caller ID prefixes. If John Doe called Alleyne Inc., the caller ID gets modified
by VoIP.ms so that the name is "[AI] John Doe <X>" (where X is John's number).
The "[AI] " part tells us visually in the caller ID what line he called into.

Presumably, you could use that information to select a ring tone as well, but
I'm not sure that'd be easy to do on your average SIP client. If you're going
through Asterisk/FreeSWITCH, you'd have more opportunity to mess with the Caller
ID for sure. I'd imagine some SIP clients could have different ring tones for
different caller IDs, but not sure how flexible.

> I'd like an answering machine function with multiple mailboxes for various
> family members.

The VoIP.ms service does this easily with it's subaccounts. You can also
definitely do this with an Asterisk or FreeSWITCH server of your own.

> I'd like to be able to access those messages (or at least be
> able to see that there are pending messages) from an computer.

With VoIP.ms at home and FreeSWITCH at work, I get copies of voicemail messages
by email, with a WAV or AIFF file attached. That makes it easy to listen to
messages.

To manage (i.e. delete) messages, I still dial in to a voice menu. There was a
web interface I was playing around with for FreeSWITCH, but didn't spend enough
time to get it working.


> Some of these may not be possible/easy, but that's what I'd *like*.
> 

The cost/benefit I'd recommend is to try VoIP.ms, and see how far you can get
with just their services and SIP clients. You'd definitely have more power and
flexibility with your own Asterisk/FreeSWITCH server, but it takes a lot more
work to understand, setup, and manage.

I'm happy with just VoIP.ms for family stuff. To satisfy requirements at work,
FreeSWITCH made more sense.


> I currently have the 416 number via Primus which connects to a VOIP box (a Cisco
> SPA-122 that Primus provided).

You can port existing telephone numbers to VoIP.ms like you would any other
provider. I've ported several Bell numbers to VoIP.ms.

> Any recommendation for a cheaper/better supplier
> for the multiple numbers I want to have?
> 

I'd definitely recommend VoIP.ms.

I'd heard good things about another provider back cira 2010/2011 when I was
looking... Callcentric I think? I've never used them though. VoIP.ms has worked
well -- very inexpensive, and with SIP, fully interoperable with free software
options. For technical users though.



More information about the talk mailing list