[simanu]

Hello, I have several DID numbers registered in different
countries and cities and was wondering whethere anyone
knew of a service (free and online) where one could
register one DID number and have it ridirected to another
SiP address.

This is because I'm about to move from the
x-Pro softphone (which can have 10 different line appareances
set up) to the new SIPURA 841 hardphone, which "only"
has 4 appareances.

What I'd like to do, and don't know if it's possible,
is have 3 or more DID numbers (from different countries)
redirected to the one DID I use the most so I wouln't have
to use up too many line appareances.

Is this possible?

On the same topic, I've tried to ring up a SIP number
using the sip syntax userid@domain using x-PRo and
could never get through. Getting "no dns left" error.

Thank you for the help,
manny

[mberlant]

You will have to use Call Forwarding to consolidate your inbound services and Dial Plan management to be able to exploit the outbound capabilities of each service.

A good example of how to integrate FWD with BV is here. Follow nabeelj's instruction for forwarding your FWD to BV and my instruction for what Dial Plan element to add for integrated outbound calling.

It might help if you told us what 10 services you have, including which ones you use for inbound only, which for outbound only and which for bothway calling. Then, we may be able to advise you better on which services to combine on each of your four lines.

Of course, it sounds like you are dangerously near the critical mass that will push you to set up a VoIP PBX, like asterisk, in your near future.

[simanu]

Hi, and thanks for the reply.

So, clarifying what I meant... I am currently using X-Pro to receive and place calls using several sip providers. Soon I'll recieve the new Sipura 841 phone with 4 appearances and will have to cut down my sip services to 4.

My SIP services are:

->Incoming:
1) www.freenumber.it, Rome number
2) www.freenumber.it, Turin number
3) www.freenumber.it, Milan number
4) www.stanaphone.com, New York number
5) www.sipgate.de, German local number
6) UK geographical number with FTW sister company (already forwarding to FWD)


<- outgoing
1) www.stanaphone.com
2) www.sipphone.com (can get rid of this)
3) www.austechpartnerships.com/atp/ (for local calls in Australia)
4) www.squillo.it (mainly to call 1800 numbers in Italy)
5) www.mutualphone.com
6) www.babble.net
7) FWD (to call other sip users and 1800 numbers)

What I would like to do, while keeping all DID numbers, is keep at least the following outgoing services:
stanaphone, babble and Squillo.

As for the incoming numbers ideally I would have all DIDs forwarded to one of the freenumber.it numbers (the service with the best sound quality). I have no idea if this is possible, though.

Sorry if my question sounds silly, but I'm a newbie in this voip world!

many thanks,
manny

[mberlant]

Your clarification is perfect. I need to do some research in order to give you solid advice.

While you are waiting you might want to rummage around the asterisk forum, just in case your desires are too much for the SPA-841.

[simanu]

Gee, thanks so much for your help! You're a star!

I will read (and have been reading) stuff about asterisk but honestly it all pretty much sounds like double dutch to me (no offence to my dutch friends out there!).

Let me know if you need help with the websites in Italian.

Thanks again,
manny

[edgreenberg]

You are a definite candidate for an asterisk server. This would allow you to have all your inbound sip services arrive at a central point, fall through a dial plan that would make them ring on your phone, or go to voicemail, or go to a menu tree, or whatever you wanted.

Then you could manage your outbound calling by designing a dialplan to do the appropritate least-cost routing.

You can do an Asterisk PBX on any reasonable linux machine.

If you are enough of a techie to have all the service you describe above, I guarantee you will have a fun time implementing this.

[simanu]

Hey, thanks for the encouragement!

I have been doing some studying and I am not sure I understand
100% what Asterisk can do for me, or better, how it can do it.

let's see...

First of all, I would personally opt for Asterisk@home, sounds like the most
suitable package for me, a linux-virgin.

I will need:

1) an old computer (at least Pentium II and 300 MhZ) with an network card.
I assume I don't need anything else but the basic parts of a computer;
no soundcard, no other fancy stuff, no huge harddisk and so on.
Of course I'll need a CD drive!

QUESTION: a fast ethernet card is ok?

2) a copy of the asteriskathome.iso file on cd

3) a multi-port router/adsl modem: I have a 4-port Billion BIPAC 5100

4) a network cable

QUESTION: which type? Crossover or direct connection?

5) an IP phone. I'm getting the new SIPURA SPA 841 with 4 appearances

QUESTION: by choosing the use Asterisk am I correct to think that I
actually didn't need a 4 appearances phone? I mean, if Asterisk can
handle all the traffic a simple Budgetone could have done the job
just as well?


Now, let's say I installed Asterisk@home and it all worked (wishful thinking?),
how easy is it to set it up, and configure?


Anyways, here comes the point where I'm not sure I get how to
proceed.

- I have my Asterisk BOX connected to the internet and always on. RIGHT?
- I have my SIPURA SPA 841 connected to the internet (using another port
of the 4-port modem/router)

- I configure my SIPURA to "register" with my Asterisk box and NOT
with all my SIP providers. The Asterisk Box will in fact register with
all my SIP services. RIGHT?


That's it? Now, depending on my Asterisk configuration and my dial plans
all I do is pick up the SIPURA phone and dial a number and the Asterisk box
will route it the right way, according to my settings? RIGHT?

THis applies to both INCOMING and OUTGOING services?


I have 7 DID numbers. Al I would have to do is setup the Asterisk box
to register with all 7 DID services and then, when anyone of the DID numbers receives
a call Asterisk will direct it to my SIPURA phone. RIGHT?

Is this is how it works, does Asterisk also "direct" the actual Caller ID (I mean,
caller phone number), or my SIPURA phone will show "Asterisk"?


I am truly excited to get all this up and running and I can't thank you guys enough
for helping me out!


If all I've said above is totally incorrect, could anyone please point me
in the right direction? What I need is described in a message I sent previously.

Thanks so much,
Manny

[edgreenberg]

Hi Manny,

You might want to take this over to the Asterisk forum for ongoing discussion. I'm not sure how sensitive the people on this forum are to topic drift.

To answer your questions:

1. A sound card is required. Other than that, an old computer will be fine. I did this on a Pentium. 233 MHz. Later, I upgraded to a Pentium II 350 MHz. which is what I'm using now. There is one asterisk feature that requires special hardware and that is the MeetMe conference. To use that, you have to have at least one Digium Zapata card in the machine to provide timing. This is a requirement that will eventually go away.

2. I've never used Asterisk At Home. Just the base Asterisk. If you are truly a Linux Newbie, you can try AAH, but you will have a smaller community from which to get support. Either approach will work. AAH or regular Asterisk. There's actually a pretty good script for regular Linux that does an Asterisk download, compile, install. Your choices.

3. Just hang the new Asterisk box on the home network along with everything else

4. If you are plugging into a hub from a computer, a regular cable is correct.

5. You don't need four appearances. Pick the phone you like the best. One or more appearances will do. I like at least two, but this is personal preferences. Remember though, appearances have nothing to do with the amount of numbers, just how many you want to be talking to at once. You could do all this with a single line phone like an IAXy.

More in next response...

[edgreenberg]

> Now, let's say I installed Asterisk@home and it all worked (wishful
> thinking?), how easy is it to set it up, and configure?

You need to (a) define all the SIP peers in sip.conf, (b) define the phone in sip.conf, (c) design a dial plan that allows you to pick which outgoing path to choose based on the country and city code (d) design a dial plan that routes the incoming calls to the phone and then to voicemail.

>
>
> - I have my Asterisk BOX connected to the internet and always on. RIGHT?
> - I have my SIPURA SPA 841 connected to the internet (using another port
> of the 4-port modem/router)
>
> - I configure my SIPURA to "register" with my Asterisk box and NOT
> with all my SIP providers. The Asterisk Box will in fact register with
> all my SIP services. RIGHT?

All the above is correct.
>
>
> That's it? Now, depending on my Asterisk configuration and my dial plans
> all I do is pick up the SIPURA phone and dial a number and the Asterisk
> box will route it the right way, according to my settings? RIGHT?
>
> THis applies to both INCOMING and OUTGOING services?
>
>
> I have 7 DID numbers. Al I would have to do is setup the Asterisk box
> to register with all 7 DID services and then, when anyone of the DID
> numbers receives a call Asterisk will direct it to my SIPURA phone.
> RIGHT?

All of the above is also correct.

>
> Is this is how it works, does Asterisk also "direct" the actual Caller ID
> (I mean, caller phone number), or my SIPURA phone will show "Asterisk"?

If caller ID comes in, Asterisk will redirect it. I've never seen "Asterisk" as caller ID. Only the inbound number.

The only thing you ABSOLUTELY MUST do is go over to www.voip-info.com and to the various forums here, and validate that there is published successful experiences using each of your SIP services with Asterisk. Some paid services only want you to use THEIR box. A good rule of thumb is that if a generic softphone like X-Lite will work, you can make Asterisk work, but I'd hate for you to do all this work and be disappointed, so do your homework first.

One last thing: If you want to dial into an Asterisk box by FWD, feel free to call me at 522838 and ring Extension 201 or 202.

Best,
</edg>

[mberlant]

I just have two points to add here. First of all, the Asterisk@home package is just an all-in-one .iso image that wipes out the hard disk and installs a self-configuring Linux and a pre-configured-with-demo-configuration Asterisk. Beyond that it's real Asterisk on a real Linux box, so there's no special support needed for this particular 'version'.

The other point is that Asterisk does not support STUN. This means that you must place the Asterisk in public internet space or map all of the appropriate SIP and RTP ports into your Asterisk box from the outside world.