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Steve Edwards wrote:
<blockquote cite="mid:Pine.LNX.4.64.0907291934540.20942@fs.sedwards.com"
type="cite">
<pre wrap="">On Wed, 29 Jul 2009, Myles Wakeham wrote:
</pre>
<blockquote type="cite">
<pre wrap="">I have setup an Asterisk system for my home & home office.
</pre>
</blockquote>
<pre wrap=""><!---->
[snip]
</pre>
<blockquote type="cite">
<pre wrap="">The cost of all these lines with analog carriers was getting ridiculous,
so I'm moving over to a SIP carrier. I created one account for a single
phone number with a SIP carrier (BroadVoice)
</pre>
</blockquote>
<pre wrap=""><!---->
[snip]
I've never used BroadVoice, so I have nothing good or bad to say about
them. I've used Vitelity.net for several years and am pleased with them.
I have a "nominal monthly fee, pay per minute" account. They get $1.49 a
month for a DID and $0.0144 per minute. You'd have to use about 2,600
minutes (about 44 hours) before it would cost as much as a $40 per month
"analog." They have an "unlimited" inbound for $7.95 a month.
</pre>
<blockquote type="cite">
<pre wrap="">I started the process today to get our other phone numbers moved over to
BroadVoice.
</pre>
</blockquote>
<pre wrap=""><!---->
[snip]
Vitelity.net charges $18 per number ported. I've never done this.
</pre>
<blockquote type="cite">
<pre wrap="">My approach is to have one trunk provided by the SIP provider. All
numbers are allocated to that trunk (BroadVoice let me do that when I
setup the number transfer). Asterisk receives an incoming call on that
trunk and determines the calling number that it was requesting (not sure
how to get this, but Broadvoice assured me I could). Anyway after
determining what the call was destined for, I then route the call to the
appropriate context in the extensions to handle it.
</pre>
</blockquote>
<pre wrap=""><!---->
The calls should be delivered with the DID (aka DNIS, DDI, etc). Usually
you pick this up as the ${EXTEN} in your dialplan and go from there.
[snip]
</pre>
<blockquote type="cite">
<pre wrap="">Broadvoice, however, won't let me change the outgoing caller ID.
Apparently they have to do this on a trunk by trunk basis. So if I want
to have an outgoing call go through line 1 (let's say its ACME Inc), I
want it to show 'XXX-XXX-XXXX Acme Inc' for the Caller ID.
</pre>
</blockquote>
<pre wrap=""><!---->
[snip]
Being able to specify the caller ID number depends on the carrier.
Vitelity.net does. Specifying the caller ID name is not going to work. The
way it works (from 40,000 feet) is that the name is not passed onto the
"real" telephone system. The carrier for the dialed number looks up the
number in a database and presents that to the dialed number. If you dial
another VOIP account (<a class="moz-txt-link-abbreviated" href="mailto:sip:john-smith@example.com">sip:john-smith@example.com</a>) your caller ID name
should be passed.
</pre>
<blockquote type="cite">
<pre wrap="">Does this sound right? Should I have purchased all separate trunks up
front and then have the phone number transfer associated with the trunk
for it? Or is this only something that will affect outgoing calls, so
its not a big deal? And what about when the line is busy? How is that
handled? I was on the phone yesterday when another call came in, and it
came in, jumped to a different extension and then eventually went to
voice mail as I didn't answer it. Will my plan to use one trunk for all
incoming lines make sense here, or am I likely to get all of this mixed
up with calls coming in for one business and being routed to the wrong
place?
</pre>
</blockquote>
<pre wrap=""><!---->
I'm more comfortable with the word "account" than "trunk." You can have
multiple DIDs numbers associated with the same account. Some providers
make you specify (via their web site) where you want the calls to go. Some
make you configure your Asterisk server so it "registers" with their
server. I prefer registration because it let's me change things around
easier.
</pre>
</blockquote>
I had this issue with Teliax. Basically with SIP, Teliax could not (or
the protocol won't let you) set your outbound caller ID via Asterisk.
Caller ID is set on a per account basis with Teliax when using SIP(IAX
was not working well for me with Teliax). So I have two outbound pay
per minute accounts with them. One for our home use and one for my
business. I use 51 prefix for home outbound calls and 52 prefix for
business outbound calls. Then my dialplan selects the proper account
at Teliax and you get the proper caller id set. <br>
<br>
My inbound is still pots lines from the telco, btw. There is no
significant cost savings on inbound for telco vs VoIP here.<br>
<br>
Lyle Giese<br>
LCR Computer Services, Inc.<br>
<br>
<br>
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