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DSL > Asterisk > W/ existing DSL # ?

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Anonymous
May 12, 2004 12:38:35 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Is this possible?

If my phone can detect an incoming call, why can't my Asterisk box do the
same with a connection into my ethernet port? Does a simple inexpensive
device exist that can translate incoming calls through my DSL line, to my
ethernet port, to Asterisk?

My goal is to send and receive soft calls through my PC, use my existing #,
and accept multiple calls.

Is this possible with Asterisk and a DSL line?

Thank you


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Anonymous
May 12, 2004 7:43:42 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

David Wiltkins wrote:
> Is this possible?
>
> If my phone can detect an incoming call, why can't my Asterisk box do
> the same with a connection into my ethernet port? Does a simple
> inexpensive device exist that can translate incoming calls through my
> DSL line, to my ethernet port, to Asterisk?
>
> My goal is to send and receive soft calls through my PC, use my
> existing #, and accept multiple calls.
>
> Is this possible with Asterisk and a DSL line?
>
> Thank you


David,

You havn't descibed your system well but as a generic answer, YES asterisk
can do that but...

It depend on firewall configuration, asterisk configuration and you havn't
describe that !


/Mats
Anonymous
May 13, 2004 1:57:00 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

"Mats K" <kalle.kula.no.sp_m@ugg-lan.com> wrote in message
news:40a22a4f@news.broadpark.no...
> David Wiltkins wrote:
> > Is this possible?
> >
> > If my phone can detect an incoming call, why can't my Asterisk box do
> > the same with a connection into my ethernet port? Does a simple
> > inexpensive device exist that can translate incoming calls through my
> > DSL line, to my ethernet port, to Asterisk?
> >
> > My goal is to send and receive soft calls through my PC, use my
> > existing #, and accept multiple calls.
> >
> > Is this possible with Asterisk and a DSL line?
> >
> > Thank you
>
>
> David,
>
> You havn't descibed your system well but as a generic answer, YES asterisk
> can do that but...

>
> It depend on firewall configuration, asterisk configuration and you havn't
> describe that !
>
>
> /Mats
>
>

Ok, just getting started with pbx etc.. it's a little overwhelming, all
sorts of new acronyms.

I'm running Red Hat 9 and one of the RPM's for Asterisk from voip-info.org.
I am not running a traditional firewall, just a Linksys router. Asterisk
seemed to fire up fine after installing.

In the quick start guide at voip-info.org, it was recommended that I bypass
my router and connect my RH9 box directly to my DSL modem. After that, it
said to copy the default settings from one of the config files for my PHONE
and then call it's extension (default 1000).. I don't have a phone though. I
prefer to run everything over ethernet from my Asterisk server to PC and
make and recieve software calls only. This way I can integrate my existing
front end gui with asterisk.


I could not find any resources for this particular setup.

Any suggestions or pointers would be greatly appreciated!

Thanks
Related resources
Anonymous
May 13, 2004 1:30:26 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

"David Wiltkins" <not@anaddress.nospam.com> wrote in message
news:325eb49d87dafedcfb7a9c10fc70f352@news.teranews.com...
>
> Ok, just getting started with pbx etc.. it's a little overwhelming,
all
> sorts of new acronyms.
>
> I'm running Red Hat 9 and one of the RPM's for Asterisk from
voip-info.org.
> I am not running a traditional firewall, just a Linksys router.
Asterisk
> seemed to fire up fine after installing.
>
> In the quick start guide at voip-info.org, it was recommended that I
bypass
> my router and connect my RH9 box directly to my DSL modem. After that,
it
> said to copy the default settings from one of the config files for my
PHONE
> and then call it's extension (default 1000).. I don't have a phone
though. I
> prefer to run everything over ethernet from my Asterisk server to PC
and
> make and recieve software calls only. This way I can integrate my
existing
> front end gui with asterisk.

Er... If I'm correct in understanding your request then it can not be
done...

You need to terminate your DSL in a filter in order for * to access your
phone line...

Minimum requirements for your set-up will then be...

-- DSL modem -- ethernet
-- DSL -- filter <
-- Phone line -- POTS

The POTS device can be one of the following:
http://www.asterisk.org/index.php?menu=hardware

The low cost Wildcard X100P is interchangeable with a Intel 537
(AMI-IA92 / AMI-IE92) Winmodem based card and should be available for
around $25 + $30 however this is an old card and the TDM400P is the
latest technology expandable to 4 lines (FXO or FXS).

Regarding the FireWall... I've got my * server behind a FireWall with no
problems, connecting to it with a soft/hard phone from the outside and
peering with FWD with no problems also. All you need to know is what UDP
ports go where and make the necessary adjustments in your NAT/FireWall
settings and in the SIP.CONF..

.... sip.conf ...
port=5060
bindaddr=192.168.0.200
localnet=192.168.0.0/255.255.0.0
localaddr=192.168.0.200
fromdomain=server.domain.com
externip=xxx.xxx.xxx.xxx
....

.... rtp.conf ...
rtpstart=10000
rtpend=20000
....

From the above I have rules mapping UDP port 5060 and ports 10000-2000
to the internal IP address of my * server (192.168.0.200).

I would recommend that you check out the latest version from CVS since
it contains substantial enhancements for services like VoiceMail and
Internationalisation.

Anyway, all you ever wanted to know about * can be found here...
http://www.voip-info.org/wiki-Asterisk


-- Soren
Anonymous
May 13, 2004 1:30:27 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

"Soren Rathje" <soren%lolle.org@spam.me> wrote in message
news:40a32413$0$294$edfadb0f@dread11.news.tele.dk...
> "David Wiltkins" <not@anaddress.nospam.com> wrote in message
> news:325eb49d87dafedcfb7a9c10fc70f352@news.teranews.com...
> >
> > Ok, just getting started with pbx etc.. it's a little overwhelming,
> all
> > sorts of new acronyms.
> >
> > I'm running Red Hat 9 and one of the RPM's for Asterisk from
> voip-info.org.
> > I am not running a traditional firewall, just a Linksys router.
> Asterisk
> > seemed to fire up fine after installing.
> >
> > In the quick start guide at voip-info.org, it was recommended that I
> bypass
> > my router and connect my RH9 box directly to my DSL modem. After that,
> it
> > said to copy the default settings from one of the config files for my
> PHONE
> > and then call it's extension (default 1000).. I don't have a phone
> though. I
> > prefer to run everything over ethernet from my Asterisk server to PC
> and
> > make and recieve software calls only. This way I can integrate my
> existing
> > front end gui with asterisk.
>
> Er... If I'm correct in understanding your request then it can not be
> done...
>
> You need to terminate your DSL in a filter in order for * to access your
> phone line...
>
> Minimum requirements for your set-up will then be...
>
> -- DSL modem -- ethernet
> -- DSL -- filter <
> -- Phone line -- POTS
>
> The POTS device can be one of the following:
> http://www.asterisk.org/index.php?menu=hardware
>
> The low cost Wildcard X100P is interchangeable with a Intel 537
> (AMI-IA92 / AMI-IE92) Winmodem based card and should be available for
> around $25 + $30 however this is an old card and the TDM400P is the
> latest technology expandable to 4 lines (FXO or FXS).
>
> Regarding the FireWall... I've got my * server behind a FireWall with no
> problems, connecting to it with a soft/hard phone from the outside and
> peering with FWD with no problems also. All you need to know is what UDP
> ports go where and make the necessary adjustments in your NAT/FireWall
> settings and in the SIP.CONF..
>
> ... sip.conf ...
> port=5060
> bindaddr=192.168.0.200
> localnet=192.168.0.0/255.255.0.0
> localaddr=192.168.0.200
> fromdomain=server.domain.com
> externip=xxx.xxx.xxx.xxx
> ...
>
> ... rtp.conf ...
> rtpstart=10000
> rtpend=20000
> ...
>
> From the above I have rules mapping UDP port 5060 and ports 10000-2000
> to the internal IP address of my * server (192.168.0.200).
>
> I would recommend that you check out the latest version from CVS since
> it contains substantial enhancements for services like VoiceMail and
> Internationalisation.
>
> Anyway, all you ever wanted to know about * can be found here...
> http://www.voip-info.org/wiki-Asterisk
>
>
> -- Soren
>
>

Thanks for the reply. I'm getting more and more confused the more I read
about this. Are you saying that my machine cannot "listen" for a phone call
connect either directly to my linksys router OR connected directly to my DSL
modem?

If you could please bare with and answer the following questions I'd be
really grateful.

1) Is it possible to call my existing DSL phone number (issued by the phone
company) and have Asterisk answer? Or do I need to go through a third-party
company?

2) Is it possible for me to call anybody, anywhere in the US or do I need to
go through a service?

3) Can my single line accept multiple calls even if I'm on the phone? As in
I'm talking to someone, another one calls and is put into voice mail, and
then another one calls and is put into voice mail etc.. What happens if 10
people call at once? Can Asterisk queue all 10 calls until I can transfer
over to them?

4) I was a little thrown off by your diagram. You mentioned your Asterisk
machine is behind a firewall, but your using a filter.. is this what you
mean?

DSL line > DSL modem > router/firewall > asterisk box
DSL line > Filter ------phone line----------wildcard etc > asterisk box




Any clarification would be greatly appreciated.


















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Anonymous
May 13, 2004 2:49:59 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

"David Wiltkins" <nowhere@nothere.notthere> wrote in message news:<94d2850a9dcdb3f2c09f574a17dc495d@news.teranews.com>...

> 1) Is it possible to call my existing DSL phone number (issued by the phone
> company) and have Asterisk answer? Or do I need to go through a third-party
> company?
>

Which DSL-phone number??? You just get a IP ...

> 2) Is it possible for me to call anybody, anywhere in the US or do I need to
> go through a service?
>

You can call any VoIP-phone, but for POTS you need a service for
terminating.

> 3) Can my single line accept multiple calls even if I'm on the phone? As in
> I'm talking to someone, another one calls and is put into voice mail, and
> then another one calls and is put into voice mail etc.. What happens if 10
> people call at once? Can Asterisk queue all 10 calls until I can transfer
> over to them?
>

That's just a question of your bandwith. Calculate 20 KBit/s for G.729
and 160 KBit/s for G.711.

> 4) I was a little thrown off by your diagram. You mentioned your Asterisk
> machine is behind a firewall, but your using a filter.. is this what you
> mean?
>

I think you confused each other. I assume filter means splitter (a
little box which separates the frequencys for POTS and DSL into two
lines).

is this your setup?

----> POTS (PlainOldTelephoneSystem)
Copper-Line ---> Splitter
----> DSL-Modem ----> Router ----> Asterisk-box

The VoIP-docs mean that you directly connect your broadband modem to
the network card in your * box and use use the * box as router. But it
should also work with you hardware router by using port-mapping.

By the way, if it's a Linksys WRT-54G you can get the GPL'ed
Linux-firmware. There's even a slim Debian distribution available. The
router should be to weak, but it would be great If one could install
Asterisk directly on the router ;-)

Juhura
Anonymous
May 14, 2004 12:30:32 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

> 1) Is it possible to call my existing DSL phone number (issued by the phone
> company) and have Asterisk answer? Or do I need to go through a third-party
> company?

By DSL phone number, you mean the phone number of the analog phone line
upon wich DSL is running? You will need an FXO device connected to the
filtered analog phone line. The easiest way to get an FXO is to buy a
card like:
http://www.digium.com/index.php?menu=devkit-fxofxs
from Digium. This card also has an FXS to connect to a hard phone. This
is a good idea to have anyways. On my system at home, I always have the
house line connected to an FXS device and my analog line to the FXO card.
I also have a bypass switch in case I really botch things or we lose power.

You do not need special service for the single line. You can contract with
iconnecthere.com (and others) to get a new phone number (about $9/mo) that
will be send calls directly to your asterisk box. I doubt there is any number
portability available. This might be a way to be able to accept multiple phone
calls, though they might limit that ability to business customers. That and
your DSL connection probably does not have enough uplink bandwidth to support
more than one or maybe two calls (unless you use G.729 or something like that).

For analog calls though, 1 line, 1 call in or out.


>
> 2) Is it possible for me to call anybody, anywhere in the US or do I need to
> go through a service?

You can call anyone on FreeWorldDial for free. If your friends have asterisk
systems or have softphones and their firewalls set up properly, you can make
them extensions of your system (or you of theirs..). Beyond that, you need a
service such as iconnecthere.com or nuphone.com. I use these two for US, local
long distance, and international calling.


>
> 3) Can my single line accept multiple calls even if I'm on the phone? As in
> I'm talking to someone, another one calls and is put into voice mail, and
> then another one calls and is put into voice mail etc.. What happens if 10
> people call at once? Can Asterisk queue all 10 calls until I can transfer
> over to them?

No. You either need an expensive, high capacity T1/PRI line or multiple
analog phone lines.

>
> 4) I was a little thrown off by your diagram. You mentioned your Asterisk
> machine is behind a firewall, but your using a filter.. is this what you
> mean?

The filter is used to prevent all the DSL signals from getting to the phone.
It's usually a little box sitting outside the house or you might have one
for every phone in your house. It's not a filter in the firewall/router
sense.

-- John
Anonymous
May 14, 2004 3:38:55 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

David Wiltkins wrote:
> "Soren Rathje" <soren%lolle.org@spam.me> wrote in message
> news:40a32413$0$294$edfadb0f@dread11.news.tele.dk...
>> "David Wiltkins" <not@anaddress.nospam.com> wrote in message
>> news:325eb49d87dafedcfb7a9c10fc70f352@news.teranews.com...
>>>
>>> Ok, just getting started with pbx etc.. it's a little overwhelming,
>> all
>>> sorts of new acronyms.
>>>
>>> I'm running Red Hat 9 and one of the RPM's for Asterisk from
>>> voip-info.org. I am not running a traditional firewall, just a
>>> Linksys router. Asterisk seemed to fire up fine after installing.
>>>
>>> In the quick start guide at voip-info.org, it was recommended that
>>> I bypass my router and connect my RH9 box directly to my DSL modem.
>>> After that, it said to copy the default settings from one of the
>>> config files for my PHONE and then call it's extension (default
>>> 1000).. I don't have a phone though. I prefer to run everything
>>> over ethernet from my Asterisk server to PC and make and recieve
>>> software calls only. This way I can integrate my existing front end
>>> gui with asterisk.
>>
>> Er... If I'm correct in understanding your request then it can not
>> be done...
>>
>> You need to terminate your DSL in a filter in order for * to access
>> your phone line...
>>
>> Minimum requirements for your set-up will then be...
>>
>> -- DSL modem -- ethernet
>> -- DSL -- filter <
>> -- Phone line -- POTS
>>
>> The POTS device can be one of the following:
>> http://www.asterisk.org/index.php?menu=hardware
>>
>> The low cost Wildcard X100P is interchangeable with a Intel 537
>> (AMI-IA92 / AMI-IE92) Winmodem based card and should be available for
>> around $25 + $30 however this is an old card and the TDM400P is the
>> latest technology expandable to 4 lines (FXO or FXS).
>>
>> Regarding the FireWall... I've got my * server behind a FireWall
>> with no problems, connecting to it with a soft/hard phone from the
>> outside and peering with FWD with no problems also. All you need to
>> know is what UDP ports go where and make the necessary adjustments
>> in your NAT/FireWall settings and in the SIP.CONF..
>>
>> ... sip.conf ...
>> port=5060
>> bindaddr=192.168.0.200
>> localnet=192.168.0.0/255.255.0.0
>> localaddr=192.168.0.200
>> fromdomain=server.domain.com
>> externip=xxx.xxx.xxx.xxx
>> ...
>>
>> ... rtp.conf ...
>> rtpstart=10000
>> rtpend=20000
>> ...
>>
>> From the above I have rules mapping UDP port 5060 and ports
>> 10000-2000 to the internal IP address of my * server (192.168.0.200).
>>
>> I would recommend that you check out the latest version from CVS
>> since it contains substantial enhancements for services like
>> VoiceMail and Internationalisation.
>>
>> Anyway, all you ever wanted to know about * can be found here...
>> http://www.voip-info.org/wiki-Asterisk
>>
>>
>> -- Soren
>>
>>
>
> Thanks for the reply. I'm getting more and more confused the more I
> read about this. Are you saying that my machine cannot "listen" for a
> phone call connect either directly to my linksys router OR connected
> directly to my DSL modem?
>
> If you could please bare with and answer the following questions I'd
> be really grateful.
>
> 1) Is it possible to call my existing DSL phone number (issued by the
> phone company) and have Asterisk answer? Or do I need to go through a
> third-party company?

What do you mean with DSL phone ?

Is this a analogue telephone or an IP telephone ?
Maybee you could post a url to the DSL providers product page and maybee you
could tell us what type of DSL phone you have.

But to clarify a bit, no 3rd parties ar needed if there is an analogue or IP
telephone we are talking about.
If it's an analogue telephone then you have to buy a card to the PC with
asterisk.
If it's an IP telephone then you just have to configure asterisk to talk to
that provider, there are several examples on "Free World Dialup"
configurations that you should be able to follow.


> 2) Is it possible for me to call anybody, anywhere in the US or do I
> need to go through a service?

NO you don't need a service if its an analogue telephone or a IP telephone
service that asterisk can interface with.


> 3) Can my single line accept multiple calls even if I'm on the phone?
> As in I'm talking to someone, another one calls and is put into voice
> mail, and then another one calls and is put into voice mail etc..
> What happens if 10 people call at once? Can Asterisk queue all 10
> calls until I can transfer over to them?

It depends.

> 4) I was a little thrown off by your diagram. You mentioned your
> Asterisk machine is behind a firewall, but your using a filter.. is
> this what you mean?
>
> DSL line > DSL modem > router/firewall > asterisk box
> DSL line > Filter ------phone line----------wildcard etc > asterisk box

The filter is a physical box where the cable is split into a DSL connection
and a telephone connection. The method used is that the telephone connection
uses one frequency range and the DSL uses another on the same physical
cable, so you can say they share the cable !

> Any clarification would be greatly appreciated.


Oh by the way the analogue telephone could be a ISDN telephone as well but
from what I know havn't ISDN penetrated the USA market in a significant way.


So as I have said earlier as mutch details as possible avoids a lot of
confusion, questions and speculations !


Kind regards
Mats
Anonymous
May 14, 2004 5:05:52 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

"David Wiltkins" <nowhere@nothere.notthere> wrote in message
news:94d2850a9dcdb3f2c09f574a17dc495d@news.teranews.com...
>

1)
If you have a phone number where you can connect a physical (or soft)
phone - YES, Asterisk can handle the call with the proper interface.

2)
You will most likely need a service somewhere to terminate your calls on
regular phone lines if that is what you want.
http://www.voip-info.org/tiki-index.php?page=VOIP+Servi...

3)
PSTN - NO. Any call waiting, voicemail on your phone number is handled
by your PTT and is not even getting close to your set of wires if you
are busy on the phone.

ISDN - YES. ISDN will provide from 2 (BRI) to 30 (PRI) lines that can be
handled by Asterisk in the way you describe.

4)
YES, the filter is for me to split DSL and PSTN into two sets of cables.
http://computer.howstuffworks.com/dsl.htm

-- Soren
Anonymous
May 14, 2004 5:05:53 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Soren Rathje wrote:

> "David Wiltkins" <nowhere@nothere.notthere> wrote in message
> news:94d2850a9dcdb3f2c09f574a17dc495d@news.teranews.com...
>
>...
> 3)
> PSTN - NO. Any call waiting, voicemail on your phone number is handled
> by your PTT and is not even getting close to your set of wires if you
> are busy on the phone.
>
> ISDN - YES. ISDN will provide from 2 (BRI) to 30 (PRI) lines that can be
> handled by Asterisk in the way you describe.
>...

Now I'm getting confused. I didn't think that it was possible to have DSL
over an ISDN line. Have I been missing out?

-- John
Anonymous
May 14, 2004 5:41:22 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

John Sutter wrote:
> Soren Rathje wrote:
>
>> "David Wiltkins" <nowhere@nothere.notthere> wrote in message
>> news:94d2850a9dcdb3f2c09f574a17dc495d@news.teranews.com...
>>
>> ...
>> 3)
>> PSTN - NO. Any call waiting, voicemail on your phone number is
>> handled by your PTT and is not even getting close to your set of
>> wires if you are busy on the phone.
>>
>> ISDN - YES. ISDN will provide from 2 (BRI) to 30 (PRI) lines that
>> can be handled by Asterisk in the way you describe.
>> ...
>
> Now I'm getting confused. I didn't think that it was possible to
> have DSL over an ISDN line. Have I been missing out?
>
> -- John

Yes it's possible to share the same cable for both ISDN and DSL.


/Mats
!