I'm using E61i. So far only played with the built-in VoIP client provided by Nokia. It works okay over wifi, but not as well as Sipura hardware (and not nearly as well as Polycom). Still, being able to make free international calls from any Starbucks isn't bad.

I checked out the iSkoot page; from a quick read it looks like they have some gateway numbers in the supported countries (e.g., an 03 number in Malaysia) and their software dials that number from your phone. That number then connects the call to the Skype neywork. Here in Malaysia, where IDD calls are cheaper than local calls (at least on prepaid), that doesn't seem like a great deal - with Digi I would pay RM0.38/minute to dial the gateway number, when I could just dial direct to Spain for RM0.18/minute.

Which model is it? Most VoIP boxes don't connect to the phone network and therefore don't require SIRIM certification. Of the Sipura/Linksys ones, only the SPA-3000 series have such a connection.

Perhaps the customs people are mistaken and don't understand the technology as well as they think.

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Added on February 8, 2008, 2:33 amAnyone stocking Linksys PAP2 in KL?

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Added on March 5, 2008, 3:17 pmAre there any companies in Malaysia or Singapore providing cheap SIP termination to those two countries (Malaysia and Singapore)?

This post has been edited by rajulkabir: Mar 5 2008, 03:17 PM

To answer my own question, with some googling I came up with www.irepublics.com. They charge RM0.10 for fixed line and RM0.16 for handphone termination within Malaysia. Not great rates but I tried the service and it seemed to work so far, so that's better than nothing. Will do a few more trials.

Not bad at all. I used it to ring my fixed line, and listened on the phone while speaking into my computer headset. The delay was barely noticeable. Ping time to their SIP server via Streamyx is about 45ms.

Yeah, they are in Malaysia. I'm not aware of any foreign providers with servers here. I think the market is probably too small and the cost of bandwidth too high.

Perhaps your SIP clients are doing a reinvite which is causing problems with your NAT. Maybe you can disable reinvites?

I went to the SMS2talk web site.

When I click on "Products" and "SMS Call", I get an error message ("Sorry. An error encountered.").

When I click on "Rates", choose my calling-from, then choose my calling-to, I get an error message ("Object cannot be cast from DBNull to other types.").

Basically it was not possible to get any useful information from the site.

If the company cannot even manage to put up a working web site (which is very easy to do) then I would not want to trust them with my telephone calls.

If there is anyone with Asterisk experience who would like to pick up a bit of extra work, please let me know. It involves providing occasional troubleshooting and reconfiguration for a local company. PM me if interested.

http://backsla.sh/betamax for price comparison of Betamax VoIP services.

Try smslisto.com - about RM0.15/minute to Saudi landlines and RM0.45/min to handphones.

Those two are part of the betamax group I mentioned in the post directly above yours.

My experience with their services:

For calls to Europe and North America, fantastic.

For calls to Australian landlines, so-so. For calls to Australian handphones, crap.

For calls to Asia, lousy - because the latency ends up being so high that echo cancellation can't work and there is an annoying delay.

When I am traveling outside of Asia, I use it to call back to Malaysia and that works fine. Also I use it to make my European and US numbers ring on my Digi phone, and that works fine too. It's just Malaysia->Asia calls that suck.