Slingshot NDSL + VOIP setup + analog phones

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Aaroona

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#64736 21-Jul-2010 18:11

The cut off date for big time is approaching fast... very fast. So What I want to know is this, because i want to move to NDSL with Slingshot.:

what device do I need to get my analog phone connected to my network? I am going to setup a network like this:

ADSL MODEM --> Gateway (PC with eBox or similar installed, providing QoS and SIP/PBX server) ---> my local network ---> SOMETHING ---> my analog phone. (only one analog phone) and from there I will also have some nokia smart phones setup.

But does anyone know what stuff I need to get this going? im thinking a Linksys adapter, but not sure if I am understanding this concept properly or not. Please help.

Also, im looking at something in the >$100 range (so obviously im happy with secondhand gear, as long as it works).

Thanks

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#354599 21-Jul-2010 20:02

Get a Linksys SPA2102

antoniosk

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#354613 21-Jul-2010 20:39

Aaroona: The cut off date for big time is approaching fast... very fast. So What I want to know is this, because i want to move to NDSL with Slingshot.:

what device do I need to get my analog phone connected to my network? I am going to setup a network like this:

ADSL MODEM --> Gateway (PC with eBox or similar installed, providing QoS and SIP/PBX server) ---> my local network ---> SOMETHING ---> my analog phone. (only one analog phone) and from there I will also have some nokia smart phones setup.

But does anyone know what stuff I need to get this going? im thinking a Linksys adapter, but not sure if I am understanding this concept properly or not. Please help.

Also, im looking at something in the >$100 range (so obviously im happy with secondhand gear, as long as it works).

Thanks

1. You'll need an Internet line of some sort of course. Telecom markets four types with a QOS-like feature - starting with UBA (no QOS) going up to EUBA180 (180kbps symmetric guarenteed, in addition to rest of line. Your ISP can confirm what they offer. Most Residential customers are ok eith EUBA40 or EUBA90 - even UBA is ok, but it's the lowest of the low.

2. You need a VOIP subscription with a provider. Like Xnet, Slingshot, Callplus or even SKYPE. Usually it's better to use the ISP's own voice service, to minimise the number of connection points. You also need to check usage policy - taking a slinghsot internet and using Xnet voice means you'll be chewing through your Internet usage as it will all just look like internet traffic. But Slingshot with Slingshot - well they shouldn't be charging you then.

3. You need a suitable box. Biddlecorp recommends the SPA2102. Most of these boxes come with 1 or 2 FXS ports on the back, for connecting 1 or 2 phones, fax machines and so on with 1 or 2 lines. They also have a 4-port switch for internet etc. Xnet have moved to an open model now, where they will supply you settings but not lock down your box...

4. VOICE FEATURES. Most important. Nearly anyone can knock up 'dialtone'. Things like Caller ID, Call Waiting, Faxability and so on - if these are important, get a Guarentee with the provider they can supply to you. Lots of people have been surprised that things like Call Waiting don't work with Voda for example (Unless they fixed it now?)

5. PRICE. Some voip providers have lovely hidden charges to make up for the cheap rates they market...

________

Antoniosk

webwat

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#355892 24-Jul-2010 19:49

I guess you are planning to setup the modem in half bridge mode to send everything to your NAT firewall? Might be more secure if you forward the SIP ports to a separate PBX box/appliance like "astlinux", since the PBX will add vulnerabilities to your firewall. Since you have a PBX, you dould look at a POTS card or USB to POTS device installed directly on the PBX. Some of the PCI POTS cards are cheap and nasty, and others are really expensive. I think the better cards tend to have hardware-based echo cancellation.

Time to find a new industry!

farcus

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#355899 24-Jul-2010 20:21

you might need to reconsider your options as it appears that Slingshot have pulled their NDSL offering.
It seems it's no longer listed as an available plan on their website.

nickb800

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#355903 24-Jul-2010 20:29

farcus: you might need to reconsider your options as it appears that Slingshot have pulled their NDSL offering.
It seems it's no longer listed as an available plan on their website.

Nope, its still up there, its just called 'broadband without homeline'. Youve gota enter your address to find out pricing.

farcus

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#355907 24-Jul-2010 20:42

nickb800:
farcus: you might need to reconsider your options as it appears that Slingshot have pulled their NDSL offering.
It seems it's no longer listed as an available plan on their website.

Nope, its still up there, its just called 'broadband without homeline'. Youve gota enter your address to find out pricing.

ahhh . . . so it is. I missed it completely.
I wonder if you still get the free italk line thrown in? It doesn't appear to be listed as one of the features any more. Unless I am missing something again.

nickb800

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#355911 24-Jul-2010 20:51

farcus:
nickb800:
farcus: you might need to reconsider your options as it appears that Slingshot have pulled their NDSL offering.
It seems it's no longer listed as an available plan on their website.

Nope, its still up there, its just called 'broadband without homeline'. Youve gota enter your address to find out pricing.

ahhh . . . so it is. I missed it completely.
I wonder if you still get the free italk line thrown in? It doesn't appear to be listed as one of the features any more. Unless I am missing something again.

Yeah good question. While im sure they would be wishing to simplify their product range, it would surprise me if they removed the free iTalk line, as it really makes their nDSL plan stack up price wise (I save ~$18 per month by going naked with voip).

Even including geeks, i cant see much of a market for a large broadband cap but no homeline. Its getting there high cellphone penetration and lower prices, but I think most people still want a homeline, whether POTS or VOIP.

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Aaroona

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#355914 24-Jul-2010 21:14

it does still come with it, JSYK.

Im thinking about going NDSL + pay an extra $$ for the upgraded VOIP plan, plus $25 for extra 20GB of data..

Just thinking about what modifications I'm going to make to my network plan, however. Might end up buying a Linksys router, put tomato on it, then run the PBX system as a separate system (like advised above).

That way, I'm not trying to find software that has a PBX built in, and then it also helps narrow down any issues. If I have issues with the firewall, then I know its the firewall, and nothing to do with VOIP (plus it means VOIP won't be down while I fix the firewall) and visa versa.

Thanks for all your suggestions guys, much appreciated. The one thing I will have to look up is astlinux. Never heard of it, but I assume by the name that its something to do with Asterisk and Linux being coupled to function as only a PBX.

I'm also looking at Askozia as a solution for my PBX system.

ChillingSilence

301 posts

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#356330 26-Jul-2010 08:15

Hi Aaroona,

Tomato / Gargoyle are great firmwares to whack onto something like a WRT54GL. Keep in mind that it won't flash on anything that has "ADSL" in it, so you've gotta have the likes of an AM300 in Half-bridge (Or a Draytek DV120). You can get a regular modem and avoid half-bridge, but you end up with double-NAT and it's not as nice a solution by any stretch of the imagination.

Basically if you setup Tomato, it's incredibly easy to get going with solid QoS, and from there you can put pretty much anything you like behind the router. ATA, Softphone, Asterisk box, take your pick.

This may help you with Tomato + QoS:
http://c2s.co.nz/screencast/voip003/
http://www.c2s.co.nz/blog/?p=16
The QoS article is a little rough and could do with tidying up, but it'll get you pointed in the right direction. Once you've got that going, it works brilliantly!!

Astlinux is nice, but if you're using generic x86 hardware, why not go for something a little friendlier such as Elastix? ;)

Hope this helps

Chill.

Aaroona

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#356428 26-Jul-2010 12:10

Thanks for all your replies, you've given me lots to think about.

I'm trying to setup Trixbox (and I've tried AsteriskNOW etc) and still can-not get inbound calls working.
Outbound is working fine, etc. but inbound will not.

When I call the number, it just says "Goodbye" and hangs up. Completely lost now.

But as far as network goes, I will be going down the WRT54GL route, (i've already got a couple of modems here to try in half bridge mode). I've got a computer here waiting to be used as the VOIP server, but like I said, unfortunately I'm having issues with inbound calls :(

ChillingSilence

301 posts

Ultimate Geek

#356432 26-Jul-2010 12:12

Make sure you have an inbound route setup. For just home use, you could use a "Catch-All" style, and not put in any CID.

Otherwise, ssh into the box and run:
asterisk -r

Then, call in, and post the log of what happens on the console.

Aaroona

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#356435 26-Jul-2010 12:15

Okay, just did that, and this is what it comes up with:

Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox1 (pid = 2555)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
    -- Executing [s@default:1] Playback("SIP/2talk-00000005", "vm-goodbye") in new stack
    -- <SIP/2talk-00000005> Playing 'vm-goodbye.gsm' (language 'en')
    -- Executing [s@default:2] Macro("SIP/2talk-00000005", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/2talk-00000005", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/2talk-00000005", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/2talk-00000005", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/2talk-00000005", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/2talk-00000005' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on 'SIP/2talk-00000005'
trixbox1*CLI>

sbiddle

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#356522 26-Jul-2010 14:02

You still don't have a correct inbound route set so Asterisk treats it as an insecure anonymous SIP connection and disconnects you with a "Goodbye" message.

Aaroona

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#356523 26-Jul-2010 14:04

sbiddle: You still don't have a correct inbound route set so Asterisk treats it as an insecure anonymous SIP connection and disconnects you with a "Goodbye" message.

oh.. I have no idea how to set one up then. I thoguht what I was doing was right.. :|

Is there anywhere you can point in the right direction please? Because I'm at a loss now lol.

ChillingSilence

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Ultimate Geek

#356528 26-Jul-2010 14:08

Under "Inbound Routes" create a new one, give it a description of something like "Catch All".

Scroll down, find your Ext, and set that as the Destination.

Hit Submit and apply all changes :)

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