VOIP Phones (SIP) in Taipei?

Anyone know where in Taipei I can buy a Voice Over IP phone? I mean one that speaks the SIP standard, not one of those Skype handsets that plug into USB on your computer. Some Taiwanese companies that make them are Soyo, WellTech, and LeadTek, but I haven’t been able to find anyone selling their gear locally.

A year ago I bought a SIP-gateway from this company in Taichung, and they have one office in Taipei:

http://www.sharetech.com.tw

This is kind of old, but I noticed that the Tsann Kuen (Can Kun) 3C store on Bade Road in the Guanghua computer area has the Art Dio IPF-2000 SIP phone.

[quote=“stralle”]A year ago I bought a SIP-gateway from this company in Taichung, and they have one office in Taipei:

http://www.sharetech.com.tw[/quote]

Unfortunately, I am not very literate in Chinese, so the webpage is not very useful to me. I was wondering what you mean with SIP gateway. Is it like a router? I am looking for something like a router with DSL modem, a connector for telephone landline, a connector for my landline phone and of course Ethernet. This device should be able to automatically route outgoing phonecalls through the internet via VoIP or landline, depending on the number I dial. It would be great if you could help.

What you’ve described is a router with a built in ATA (analog terminal adaptor). A SIP gateway is more like a server for VOIP calls and are pretty expensive. There are Linksys and D-Link routers with built in ATAs, not sure what’s available locally. Be careful what you get because some of those are hardcoded to a particular service provider (such as Vonage).

To do that you’d need an ATA with both FXS and FXO ports. The Grandstream Handytone 488 would do what you want, but it is just an ATA; you’d need a separate router too. What I do is have a two line phone where the local line is hooked to line 1 and the VOIP ATA is hooked to line 2.

[quote=“jlick”]
What you’ve described is a router with a built in ATA (analog terminal adaptor). A SIP gateway is more like a server for VOIP calls and are pretty expensive. There are Linksys and D-Link routers with built in ATAs, not sure what’s available locally. Be careful what you get because some of those are hardcoded to a particular service provider (such as Vonage).

To do that you’d need an ATA with both FXS and FXO ports. The Grandstream Handytone 488 would do what you want, but it is just an ATA; you’d need a separate router too. What I do is have a two line phone where the local line is hooked to line 1 and the VOIP ATA is hooked to line 2.[/quote]

Thanks jlick,

that is great help, I found the webpages of the device you mentioned, link is below.

grandstream.com/y-ht488.htm

Actually a router does not seem to be necessary, I already have one, and it seems like I can hook the ATA between my router and my DSL modem. What I don’t understand is, why do you need 2 phone lines. On the webside it says that the device has a PSTN Pass-through. Isn’t it possible to configure the device, so it chooses the calling method (SIP vs. landline) depending on the area code you dialed. For example, if I dial +1xxxx it chooses my SIP provider in the US, +49xxx SIP from Germany 02xxxx landline to Taipei? How many SIP providers would I be able to configure?

Okay, I just found the users manual, will have a look in there. Any ideas where I could purchase this device?

The ethernet port on the ATA would plug into one of the ports behind your router, not in between the router and the modem.

Sorry, that example was for a single-port ATA. With a FXS/FXO ATA you would only need a 1 line phone. It’s just that single port ATAs are a bit more common.

It’s been a while since I played with that model but I think the default is that you press 9 (programmable to another number) to go through the landline and otherwise it goes through the VOIP provider. If you want to go with fancier setups, one way you could do it is setup an Asterisk server on a linux box. Asterisk is an open source VOIP PBX. You could then make this server the SIP server for your ATA and then it would have the rules on where the call goes.

I believe the HT488 only supports one SIP provider. If you route through a local Asterisk box though, the Asterisk server could talk to multiple SIP providers.

I bought one in the US.

[quote=“jlick”]
The ethernet port on the ATA would plug into one of the ports behind your router, not in between the router and the modem.

Yeah, I thought about this option later. Of course that would work best this way. [/quote]

Good, that’s what I thought. Sounds like the right device for me then.

Setting up a Asterisk box would be more like shooting with cannons onto sparrows. Actually, the reason why I would like a stand alone device is that I don’t want to have to turn on a computer on for placing phone calls. I try to save some electricity, and a server being on continiously would be a waste, and booting up one just for a phone call, is too much of a hassle.

That’s what I read in between the lines of the user’s manual. But at least I now know what is the proper terminology for such a device, ATA and not SIP router or SIP gateway.

Maybe I can get someone in the U.S. or from home to send one over to me if I don’t find it in TW. Thanks again for you help.

There is a company in the US called iConnecthere that offers VOIP service for about $900 NT per month. They will give u a phone number in whatever country you wnat and offer 800 minutes of free calls to the US/Canada. And they will give u a free Linksys PAP2 device and will ship it to you anywhere in the world for free!!

Also, they accept internaltional credit cards!

I have the service and it is really nice…I set the phone number to match the same area code as my parents and then the call is free for them.

the website is

www.iconnecthere.com

Hopefully it is what u want :wink:

I recently signed up for Vonage in the US. Cost is $14.95 that includes 500 minutes worth of calls to US or Canada. works great. I bought a Vonage equipped Linksys router.

[quote=“schnoz”]There is a company in the US called iConnecthere that offers VOIP service for about $900 NT per month. They will give u a phone number in whatever country you wnat and offer 800 minutes of free calls to the US/Canada. And they will give u a free Linksys PAP2 device and will ship it to you anywhere in the world for free!!

Also, they accept internaltional credit cards!

I have the service and it is really nice…I set the phone number to match the same area code as my parents and then the call is free for them.

the website is

www.iconnecthere.com

Hopefully it is what u want :wink:[/quote]

I’ve used iconnecthere for years but could never figure out if I could get one of their phones to work here for me. Would I have to have my computer on all the time? How long ago did you sign up for their service? Do you pay by the month or year? Exactly what did you have to do once you received the phone? (free shipping anywhere in the world?! not the last time I looked into it… hm.)

Most VOIP phones or adaptors do not need you to leave your phone on. Usually the phone or adaptor will plug into your router. Sometimes the adaptor is built in to the router. If you don’t have a router you can get one for 1000 or less at any 3C store. (You really should have a router between your computer and modem anyways. Directly connecting your computer is just asking to be hacked.) Skype phones and adaptors on the other hand usually require your computer to be on to use them.

Every service is different, some with a monthly charge, some paid per-minute, etc.

I’ve used both packet8 and vonage. With both, I hooked the ethernet line into the router, and hooked a regular phone to the phone jack on the adaptor. With Vonage I then had to pick up the phone and dial an activation number that they sent me; with packet8 it was ready to go. After that, just pick up the phone and dial! Some services will make you do more complex setup, so ask to make sure the equipment they send you is preconfigured.

With Vonage I don’t have to dial an activation number. Not sure what that’s about.
I have an Arizona phone number and it’s just a matter of picking up the phone and as soon as you have a dial tone start dialing. The Typhoon weather was driving it nuts though. It’s only as good and clean as your ADSL line.

I only had to do that one time when I first got the adaptor. It’s probable that yours came preconfigured.

Anyone check out Gogo Talk ?

You can do the same thing with the new Yahoo Messenger

You can do the same thing with the new Yahoo Messenger[/quote]

I was thinking more about their Softphone service. And if TTN can be an upstream VOIP provider and give out numbers for this then it opens up a great number of fantastic opportunities.

Does anyone know if TFN/Chunghwa provide services like Junction Networks?

I bought a HandyTone-486 VoIP adaptor from a VoIP provider in Norway incl. Norwegian phone number and subscription. Easier and cheaper for family and customers to call a local number to reach me.

Now I am trying to connect it with a telephone in Taiwan. I suspect my problem is to configure the correct specifications in the VoIP adaptor for the local (Taiwanese) phone?
I have been looking high and low on internet to find the answer to my questions, without success, so now I post my problem.

I have tried 2 different phones connected to the HT-486, with same problem - a Philips PH-838 and a Align-885. I do get the dial tone, but then things goes wrong.

Problem description:
[ul]1. Calling a Taiwanese number I get the ring tone and the phone rings, but neither me nor the other side can hear anything when the call is answered. My VoIP register and bill the call.
2. Calling from a Taiwanese number give no ring tone and generates a voice prompt error message.
3. Calling a Norwegian number I do not get a ring tone, but the phone in Norway rings. Neither me nor the other side can hear anything. The VoIP provider register and bill the call.
4. Calling from a Norwegian number. No ring tone, no ringing on the 486-adapter phone.[/ul]
I suspect there are some wrong parameters in the 486 setup for the local phone specification, but I have not been able to find what the correct settings should be. Maybe someone here can help me? Thank you!

Here are the setting options, and my settings, for what I think is the problem. Please suggest otherwise if you feel it can solve my problem.

Send DTMF

This parameter controls the way DTMF events are transmitted. There are 3 ways: in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
My setting: via RTP (RFC2833)

DTMF Payload Type

This parameter sets the payload type for DTMF using RFC2833
My setting: 101

Send Flash Event

This parameter allows a user to control whether to send an SIP NOTIFY message indicating the Flash event, or just to switch to the voice channel when the user presses the Flash key. Default is NO.
My setting: NO

FXS Impedance

Selects the impedance of the analog telephone connected to the Phone port.
Options:
[ul]600 Ohm (North America)
900 Ohm
600 Ohm + 2.16uF
900 Ohm + 2.16uF
CTR21 (270 Ohm + 820 Ohm || 150nF)
Australia/New Zealand #1 (220 Ohm + 820 Ohm || 120nF)
Slovakia/Slovenia/South Africa (220 Ohm + 820 Ohm || 115nF)
New Zealand #2 (370 Ohm + 620 Ohm || 310nF)[/ul]
My setting: 600 Ohm (North America)
-I tried the CTR21 earlier, with same result as now

Caller ID Scheme

Select the Caller ID Scheme to suit the standard of different area.
Options:
[ul]Bellcore (North America)
ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
ETSI-DTMF (Finland, Sweden)
Denmark-DTMF
CID-Canada[/ul]
My setting: ETSI-FSK
(it works partially

I want to get one of these: taipeitimes.com/News/front/a … 2003279209
The firm Accton Technology says it has produced the world’s first cellphone that can use the popular Skype software for making free Web calls

You look at their website and it just says they’ve created a partnership to make a skype wifi phone, they haven’t produced it at all, and I bet it’s not a cell phone (Are any newspapers capable of getting anything right) I still want one though

EDIT: scrap that, maybe the newspaper was right and the company making it is wrong: digitimes.com/telecom/a20051108A9057.html

X3M,

Your calls are being seen and billed by the VOIP provider, so none of the issues you raised are at fault. The DTMF encoding is the only one that relates to the connection between adaptor and the VOIP server. The others all are related to the connection between the phone and the adaptor. Thus the caller id encoding relates to whether your phone would be able to display incoming caller id correctly.

Anyways, given the symptoms you described, I would guess that the problem is that your adaptor is behind a router which does IP sharing (aka NAT). In that case, you need to set up the adaptor to handle that situation. The usual way is to set it to use a STUN proxy, but you should contact your VOIP provider for specific instructions.

The other problem that relates to the “calls go through but can’t hear anything” is an incompatibility in codec. For best voice quality, the common codec is G711, but it’s possible also to use compressed codecs. You should verify with your VOIP provider that you are defaulting to the codec they expect.