«

Hi --

I'd like to extend an analog pbx station port from the office to my home.
It should work like an off prem extension, that is when I go off hook at
home, I should get pbx dial tone... when someone calls the extension, my
analog phone at home should ring .

I know there are a lot of boxes out there that do this, but they are
hundreds if not thousands of bucks... I'm wondering if there's an
inexpensive solution (as I'm paying for this on my own).

Thanks in advance!
Dave

"Henry Cabot Henhouse III" <> writes:
> I'd like to extend an analog pbx station port from the office to my home.
> It should work like an off prem extension, that is when I go off hook at
> home, I should get pbx dial tone... when someone calls the extension, my
> analog phone at home should ring .
>
> I know there are a lot of boxes out there that do this, but they are
> hundreds if not thousands of bucks... I'm wondering if there's an
> inexpensive solution (as I'm paying for this on my own).

You can try a pair of Sipura SPA-3000's. One of the canonical
examples they have in their FAQ's is how to set up the dialplan to do
a "hotline". Eg. when you pick up one phone it automatically calls
the some number. You can use that at the PBX-connected sipura to call
your remote sipura whenever a call comes in. The remote sipura would
be setup normally, with the pbx-connected one designated its outgoing
sip gateway. For a bit of added simplicity I'd be sure to get two
identical fxo/fxs units. You'll be screwing around with enough tricky
settings in the units without having to worry about learning two
totally different command sets. (There are probably other FXO/FXS
units that can do the job too. I'm only familiar with the Sipura
unit.)

I have a SPA-3000 that I use to feed a POTS line into my asterisk
(PBX). It works, but the reality of POTS lines is that it is
impossible to feed them into a VOIP system without either causing a
problem with reduced volume or if you crank the volume up, without
introducing a bit of echo. If your pbx outputs a digital signal (PRI
or BRI) this is probably the way you want to go. There are quite a
few universities that have hooked their old PBX to either Cisco
equipment or asterisk/SER and use SIP to connect things. In their
case the goal is also to be able to connect to their PBX over the net,
but only for the purpose of calling the users on their pbx. The
setups should be similar enough that the configuration examples might
prove useful.

http://www.internet2.edu/sip.edu/

-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

Hi ...

I read a review of the 3000, and it stated that as the 3000's only support
SIP, they have to register with a SIP server. The review also said that
configuring the dial plan was not for the average teleworker.

Do I have to use an outside SIP box to support these?

Thanks
Dave



"Wolfgang S. Rupprecht"
<wolfgang+ .wsrcc.com> wrote in
message news:...
>
> "Henry Cabot Henhouse III" <> writes:
>> I'd like to extend an analog pbx station port from the office to my home.
>> It should work like an off prem extension, that is when I go off hook at
>> home, I should get pbx dial tone... when someone calls the extension, my
>> analog phone at home should ring .
>>
>> I know there are a lot of boxes out there that do this, but they are
>> hundreds if not thousands of bucks... I'm wondering if there's an
>> inexpensive solution (as I'm paying for this on my own).
>
> You can try a pair of Sipura SPA-3000's. One of the canonical
> examples they have in their FAQ's is how to set up the dialplan to do
> a "hotline". Eg. when you pick up one phone it automatically calls
> the some number. You can use that at the PBX-connected sipura to call
> your remote sipura whenever a call comes in. The remote sipura would
> be setup normally, with the pbx-connected one designated its outgoing
> sip gateway. For a bit of added simplicity I'd be sure to get two
> identical fxo/fxs units. You'll be screwing around with enough tricky
> settings in the units without having to worry about learning two
> totally different command sets. (There are probably other FXO/FXS
> units that can do the job too. I'm only familiar with the Sipura
> unit.)
>
> I have a SPA-3000 that I use to feed a POTS line into my asterisk
> (PBX). It works, but the reality of POTS lines is that it is
> impossible to feed them into a VOIP system without either causing a
> problem with reduced volume or if you crank the volume up, without
> introducing a bit of echo. If your pbx outputs a digital signal (PRI
> or BRI) this is probably the way you want to go. There are quite a
> few universities that have hooked their old PBX to either Cisco
> equipment or asterisk/SER and use SIP to connect things. In their
> case the goal is also to be able to connect to their PBX over the net,
> but only for the purpose of calling the users on their pbx. The
> setups should be similar enough that the configuration examples might
> prove useful.
>
> http://www.internet2.edu/sip.edu/
>
> -wolfgang
> --
> Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
> Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

"Henry Cabot Henhouse III" <> writes:
> I read a review of the 3000, and it stated that as the 3000's only support
> SIP, they have to register with a SIP server.
....
> Do I have to use an outside SIP box to support these?

I don't believe you will need an outside SIP server in order to make a
simple "hotline" service. You only need to setup a call between the
two units and they have enough smarts in the dialplan to do that
themselves.

It is nice to have an ntp server and dhcp server so the units will set
the time automatically and set their IP addresses automatically, but
even that, I believe, is optional.

> The review also said that configuring the dial plan was not for the
> average teleworker.

The dialplan itself is only part of the joy. It's only a one-line
entry. The intimidating part is first seeing the pages and pages of
other crap that you can configure (but for the most part, don't have
to). It is not out of the realm of what someone interested in
tinkering with technical things can handle. The biggest challenge is
just not being scared away by the sheer number of entries that can be
tinkered.

-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

gratias... the Sipura units are certainly in my price range... guess it wont
hurt to try a pair.

merry christmas / happy holidays!


"Wolfgang S. Rupprecht"
<wolfgang+ .wsrcc.com> wrote in
message news:...
>
> "Henry Cabot Henhouse III" <> writes:
>> I read a review of the 3000, and it stated that as the 3000's only
>> support
>> SIP, they have to register with a SIP server.
> ...
>> Do I have to use an outside SIP box to support these?
>
> I don't believe you will need an outside SIP server in order to make a
> simple "hotline" service. You only need to setup a call between the
> two units and they have enough smarts in the dialplan to do that
> themselves.
>
> It is nice to have an ntp server and dhcp server so the units will set
> the time automatically and set their IP addresses automatically, but
> even that, I believe, is optional.
>
>> The review also said that configuring the dial plan was not for the
>> average teleworker.
>
> The dialplan itself is only part of the joy. It's only a one-line
> entry. The intimidating part is first seeing the pages and pages of
> other crap that you can configure (but for the most part, don't have
> to). It is not out of the realm of what someone interested in
> tinkering with technical things can handle. The biggest challenge is
> just not being scared away by the sheer number of entries that can be
> tinkered.
>
> -wolfgang
> --
> Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
> Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

Hi ...

I emailed Sipra tech support... I received an answer in hours...

Dear Valued Sipura Customer,

Thank you for contacting Sipura Technical Support.

Both SPA-3000's and the routers that are connected to need to have a static
WAN ip address OR so you can always know your ip address. Your routers
should be configured to allow the SIP ports 5060 and 5061 and foeward these
two on your SPA-3000's and should be "SIP friendly". Enable Make and receive
calls without registration on both SPA-3000's and set the User ID and SIP
Port on line 1 and PSTN line to unique numbers on both SPA-3000's. Enable
Voip and PSTN gateways on both spa's, spa2 - Enable VoIP Caller Auth Method:
PIN, Define the VoIP Caller 1 PIN: This PIN will be requested when you want
to gain access to the pstn line. At this point from the spa1 you can dial #2
and the spa2 answer with some beeps (asking you for the PIN) then after you
enter a correct pin, you will hear dial tone on the spa2.


***
I take it from this I have to use a PIN to access dial tone from the pbx...
is this for security?

Thanks!
Dave





"Wolfgang S. Rupprecht"
<wolfgang+ .wsrcc.com> wrote in
message news:...
>
> "Henry Cabot Henhouse III" <> writes:
>> I'd like to extend an analog pbx station port from the office to my home.
>> It should work like an off prem extension, that is when I go off hook at
>> home, I should get pbx dial tone... when someone calls the extension, my
>> analog phone at home should ring .
>>
>> I know there are a lot of boxes out there that do this, but they are
>> hundreds if not thousands of bucks... I'm wondering if there's an
>> inexpensive solution (as I'm paying for this on my own).
>
> You can try a pair of Sipura SPA-3000's. One of the canonical
> examples they have in their FAQ's is how to set up the dialplan to do
> a "hotline". Eg. when you pick up one phone it automatically calls
> the some number. You can use that at the PBX-connected sipura to call
> your remote sipura whenever a call comes in. The remote sipura would
> be setup normally, with the pbx-connected one designated its outgoing
> sip gateway. For a bit of added simplicity I'd be sure to get two
> identical fxo/fxs units. You'll be screwing around with enough tricky
> settings in the units without having to worry about learning two
> totally different command sets. (There are probably other FXO/FXS
> units that can do the job too. I'm only familiar with the Sipura
> unit.)
>
> I have a SPA-3000 that I use to feed a POTS line into my asterisk
> (PBX). It works, but the reality of POTS lines is that it is
> impossible to feed them into a VOIP system without either causing a
> problem with reduced volume or if you crank the volume up, without
> introducing a bit of echo. If your pbx outputs a digital signal (PRI
> or BRI) this is probably the way you want to go. There are quite a
> few universities that have hooked their old PBX to either Cisco
> equipment or asterisk/SER and use SIP to connect things. In their
> case the goal is also to be able to connect to their PBX over the net,
> but only for the purpose of calling the users on their pbx. The
> setups should be similar enough that the configuration examples might
> prove useful.
>
> http://www.internet2.edu/sip.edu/
>
> -wolfgang
> --
> Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
> Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

"Henry Cabot Henhouse III" <> writes:
> sipura writes:
> Both SPA-3000's and the routers that are connected to need to have a static
> WAN ip address OR so you can always know your ip address. Your routers
> should be configured to allow the SIP ports 5060 and 5061 and foeward these
> two on your SPA-3000's and should be "SIP friendly". Enable Make and receive
> calls without registration on both SPA-3000's and set the User ID and SIP
> Port on line 1 and PSTN line to unique numbers on both SPA-3000's. Enable
> Voip and PSTN gateways on both spa's, spa2 - Enable VoIP Caller Auth Method:
> PIN, Define the VoIP Caller 1 PIN: This PIN will be requested when you want
> to gain access to the pstn line. At this point from the spa1 you can dial #2
> and the spa2 answer with some beeps (asking you for the PIN) then after you
> enter a correct pin, you will hear dial tone on the spa2.
>
>
> ***
> I take it from this I have to use a PIN to access dial tone from the pbx...
> is this for security?

Adding a PIN would be a way to prevent incoming (or outgoing) calls
without the user touch-toning some a secret key. If you wanted all
calls transparently relayed to the other side, I'd think you wouldn't
really want this. Also, from a security standpoint, unless the PIN is
very long (say well over 6 digits), it isn't going to do much to stop
a computerized attack. It just isn't going to take that long for a
computer to try all 3, 4 or 5 number PINS. Trying all 6 number PINS
probably only takes a week or two.

To stop a determined attacker from making outgoing calls via your pbx
you'll want to add md5/http-digest authentication to the internet side
of the pbx-connected sipura. Just choose 32 randomly chosen letters
and numbers for your md5 password. Load that password into both of
your sipuras and you should be safe enough.

One thing I forgot to mention is there are a number of forums where
folks that like to hack their sipura's hang out. Most of them seem to
have thought about this sort of stuff for much longer than I have.

http://voxilla.com/PNphpBB2.html
http://forum.sipbroker.com/search.php?searchid=4811
http://www.dslreports.com/forum/voip

-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

> To stop a determined attacker from making outgoing calls via your pbx

Don't provide a way to do it. Seriously, if all you're doing is 'remoting'
an extension then setup a fixed dialing plan for it. Such that all
connections go to/from only the two fixed connections. You could setup the
unit at the house to understand how to treat certain calls as local. But if
this is work-related then you might just be better off using it as a second
line on a two-line phone at home. That way anyone else in the house can
pickup line 1 and use it as expected. Line 2 would be the work line and
probably only available on certain extensions.

and the best way to join a voip and pstn service is the combine-a-line...
the unit works all over the world and solves he problem of voip and pstn
ports being in different places. you acan join al your favorite single line
telco gear, like phone, answering system and even modem onto the combine
a-line and then one side to voip and the other to pstn. then you have a
powerful suite of tolls and automatic switching for your communication
set=up..

Marco
http://cgi.ebay.com/ws/eBayISAPI.dll...rd=1&sspagenam
e=STRK%3AMESE%3AIT&rd=1


"Bill Kearney" <> wrote in message
news:VNqdnZaaZpfKBzbeRVn-...
> > To stop a determined attacker from making outgoing calls via your pbx
>
> Don't provide a way to do it. Seriously, if all you're doing is
'remoting'
> an extension then setup a fixed dialing plan for it. Such that all
> connections go to/from only the two fixed connections. You could setup
the
> unit at the house to understand how to treat certain calls as local. But
if
> this is work-related then you might just be better off using it as a
second
> line on a two-line phone at home. That way anyone else in the house can
> pickup line 1 and use it as expected. Line 2 would be the work line and
> probably only available on certain extensions.
>
>

ditto

http://cgi.ebay.com/ws/eBayISAPI.dll...rd=1&sspagenam
e=STRK%3AMESE%3AIT&rd=1


"Marc Popek" <> wrote in message
news:8%Eyf.8053$ nk.net...
> and the best way to join a voip and pstn service is the combine-a-line...
> the unit works all over the world and solves he problem of voip and pstn
> ports being in different places. you acan join al your favorite single
line
> telco gear, like phone, answering system and even modem onto the combine
> a-line and then one side to voip and the other to pstn. then you have a
> powerful suite of tolls and automatic switching for your communication
> set=up..
>
> Marco
>
http://cgi.ebay.com/ws/eBayISAPI.dll...rd=1&sspagenam
> e=STRK%3AMESE%3AIT&rd=1
>
>
> "Bill Kearney" <> wrote in message
> news:VNqdnZaaZpfKBzbeRVn-...
> > > To stop a determined attacker from making outgoing calls via your pbx
> >
> > Don't provide a way to do it. Seriously, if all you're doing is
> 'remoting'
> > an extension then setup a fixed dialing plan for it. Such that all
> > connections go to/from only the two fixed connections. You could setup
> the
> > unit at the house to understand how to treat certain calls as local.
But
> if
> > this is work-related then you might just be better off using it as a
> second
> > line on a two-line phone at home. That way anyone else in the house can
> > pickup line 1 and use it as expected. Line 2 would be the work line and
> > probably only available on certain extensions.
> >
> >
>
>