Singapore PSTN configuration for VoIP ATA (e.g.:SPA3102)

[Chuckles]

What are the correct PSTN line settings for Singapore - e.g.: for use when setting up an ATA like Linksys-Cisco SPA3102 ?

- disconnect tone (480@-30,620@-30;4(.25/.25/1+2) ?)
- FXO port impedance (600 ohms?)
- tip/ring voltage (3.5?)
- on-hook speed (Less than 0.5mS?)
- Operational Loop Current Min: (10mA?)
- ring threshold (13.5-16.5V rms?)

I am using StarHub Digital Voice modem as my PSTN line (Motorola SBV5120). This is a similar post here with no answers:
http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3000-singapore-pstn-configuration-16329.html

[dao]

I don't remember doing any changes when I connect my old SPA3102 to Starhub Digital Voice modem. It works out from the box, so to say. However, I am facing the same problem as you now with Singtel Mio voice.

I will post my setting for you when I get home. Hopefully that will help you for your Starhub Digital Voice.

[Chuckles]

These settings should sound like Singapore land line tones

Dial tone: 425@-10; 10(*/0/1)
Busy tone: 425@-10; 10(0.75/0.75/1)
Disconnect tone: 425@-30,425@-30; 2(0.75/0.75/1+2)

These found at 3amsystems dot com

[Chuckles]

IDA website has a Technical Specification for Terminal Equipment connecting to PSTN

Line impedance for exchange and terminal equipment is stated: 600 ohms

[Chuckles]

Ring Frequency: 24
(source: IDA technical spec for TE connecting to PSTN)

CWT Frequency: 425@-10
(source: -10dBm at 3amsystems dot com, 425 is dial tone frequency)

[dao]

Did you manage to get it work for your starhub digital voice?

My setting is:

http://picasaweb.google.com/lh/photo/ccPoh2DuKFZPvQYhzS1VoA?feat=directlink

[Chuckles]

My SPA3102, bought in SimLimSquare, is working though my wife doesn't trust it!! :s22:

After I had learnt about all the settings, quite complex right!, the only real problem was when dialing from my home phone through the SPA3102 box to StarHub Digital Voice PSTN: calls to local numbers were not getting connected, and I hear the disconnect tone.

Changing this setting by trial and error seemed to help, though I don't know why:

PSTN Line > International Control
SPA To PSTN Gain: 3 (default was 0)

@dao: Please can you do a screenshot of the International Control settings in PSTN Line tab on your SPA3102?

[liangtam]

sooo troublesome de ah? buy the provider-supplied ATA next time.

[Chuckles]

@liangtam - you're right: provider supplied ATA are much easier to install.

But I do not want to be tied to one provider. VoIP call rate deals and call reliability keep changing so I expect to be changing service providers often.

I am trying Pfingo and Sunpage here in Singapore, also Gizmo because that has a global SIP number unlike Pfingo and SunPage which are closed to incoming SIP calls from other service providers. Also I may try some of the cheaper US ones like smartvoip, or Australia PennyTel...

[liangtam]

not all provider supplied ATA are locked sets.
Tho u shouldn't buy those excessively configurable ones if your main objective is still more on making calls.

[Chuckles]

@liangtam. True! I should have bought a Siemens IP phone from Pfingo, now that I know the configuration complexity and restriction of incoming SIP call connections with the Linksys SPA3102.

Decide in haste, repent at leisure.. :yawn:

[dao]

SPA3102 is a very powerful device, once you know how to 'play' with it. there are so many things that can be done/configured.

will post my setting when i reach home.

[dao]

Uploaded my PSTN International setting. Please use the same link, select SPA3102 album.

[Chuckles]

@dao. Thanks! Some differences with the unit I bought at Sim lim, such as impedance. Did you edit the settings? Did you buy in singapore?

[dao]

bought in SG. I might have 'cos i was playing it with asterix. Is yours working now?

[Chuckles]

@dao. Thanks for posting your info - understand you are busy. Good luck!

These PSTN settings work for SPA3102 with StarHub digital voice modem (Motorola cable model) phone jack and broadband (see link and image)

h t t p : / /expatgeeksingapore.blogspot.com/2009/07/telephone-adapter-for-pfingo-sip-calls.html


The main difference with dao settings are line impedance and SPA to PSTN gain settings

[dao]

Read your blog.

There are ways to get around the restriction of one single service in case a).

For incoming, you can have both PSTN and VOIP line registered. So, you got 2 number for incoming call to ring your phone. In your case, PSTN is your Starhub Digital Voice and VOIP is your pfingo.

For outgoing, there are more choices : the above plus 4 more VOIP registered.

[Chuckles]

@dao

You are right - I do have both a PSTN (StarHub) and a VoIP (Pfingo) line 1.

But only Pfingo users can make a VoIP/cost-free call to my Pfingo url so it is not so useful. I want another VoIP incoming line SIP URL that works for users of other VoIP service providers (like Gizmo).

On Line 1 I tried Gizmo as the VoIP service, and then set up Pfingo as one of the Gateway Accounts: however Pfingo service did not work in this configuration.

Did you manage to get SPA3102 working with Mio router and Mio Voice?

[dao]

Don't quite under what you mean "only Pfingo users can make a VoIP/cost-free call to my Pfingo url". Care to elaborate?

Actually, the PSTN line can be used for VoIP if you do not have a PSTN.

Also, I am using pfingo too. I use it as gateway3. No problem. It is Mio voice that is giving me problem using it as a gateway.

Thanks for your blog, I manage to get SPA3102 working with Mio router and Mio voice, as a PSTN. Need to spend some more time to get it working as a VoIP in SPA3102.

[Chuckles]

I have not bought a number (3xxx...). Other Pfingo users can make calls to my pfingo sip address (user-id@sip.pfingo.com): but not other voip users such as Gizmo. If this did work, this would allow free incoming global calls via internet to my home in Singapore...

With Pfingo as gw3, which voip service provider you use for Line 1?

Correction/more information:
---------------------------------
Pfingo does not require registration to make outgoing calls, so it can set up as a Gateway account with another service set as Line 1. I tested this and find it works.

=> I will try Gizmo or direct IP dialling as Line 1 service to allow free global incoming SIP uri/Sipbroker calls to my handset, and set gateway accounts to PennyTel and Pfingo for low cost IDD outgoing VoIP calls