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