Hello Members !

I have received many PMs lately and most of the users want a VOIP Thread to be a sticky but Xybirium has convinced me to create a sticky , as It would help to spread the message around.

So , we have a Dedicated VOIP Thread here in LYN itself. It would be a 'pioneer' for us all.

So you could talk about VOIP Providers here in Malaysia , the rates and anything related to VOIP.


Important Note : BEAR IN MIND , I DONT WANT ANY AGENTS TO 'SPAM' HERE . PLEASE DO NOT SPAM THIS THREAD BY SAYING 'Come and try our VOIP product -http://xyz.com'.If I ever find any posts such as that , the person who posted that would get a 3 days ban.

But if you say ' I use XXXYYYZZZ VOIP program by XYZ.SDN BHD. For more information , check out <Company's Website> , then its acceptable. As users can check out Company's Website for more details.

If you wish to promote your services , please use Garage Sales instead. You could promote your services , by giving good details and at the sametime posting your URL [Garage Sales URL]. Then its acceptable as well


Thank You

Original Message -click The Show Spoiler Button-


This post has been edited by prasys: May 21 2006, 08:49 PM

I will not subscribe it as it requires monthly subscription fee of RM10 or RM15, cant remember.

count me in if there is no monthly fees and calling from streamyx to streamyx user is free!

You say you're using PAP2. Do they supply this or you have to buy your own?

What do you mean when it is not good enough for conference call? You hear echo? noise? or just unable to make conference call?

What's their rates for calling POTs line to Singapore?

there is a monthly fee ... RM15 but I think ok lar since u can call to pstn and mobile ... save on ur std ....

fix 10 cent/min
mobile 20 cent/min
pc to pc free

I am using it now quite good no echo, noise etc

I think no ATA right now just softphone ... u want hw have to buy ur own ..

STD to Spore .... fix & mobile 0.15

I was using Jaring MY015 and I think it is the same as TMNet MY015 and also Nasioncom MY015. I understand that calling within MY015 is free even if it is Jaring or Nasioncom. Rates are okay BUT the voice quality has too much echo and also quite high latency. I am using the Linksys PAP2 by the way.

You might want to check out www.voipstunt.com or www.internetcalls.com where they provide free fixed line calls to more that 50 locations worldwide. Calls to mobiles in US, Singapore, Hong Kong and a few other locations are also FREE. The catch is you have to pay 10 Euros to allow you to make these unlimited calls.

Any comments?

I see. But, if this is softphone only, that's little difference from the existing popular Skype services, perhaps Skype even offer better quality.

I think Skype Out rates is cheaper than TmNET rates even with the currency conversion. Isn't it?

I am not sure how it sounded like over TmNET but Skype voice quality is very good on almost every Internet broadband accessible places (eg. StarBucks hotspots, hotels, SG, HK, TW and US locations) that I have use.

Have you try TMnet VoIP softphones with these USB phones?

I have bought one of these generic cheap USB phone for my laptop when travelling (no need to use the PC headsets) and they works great with Skype.

This post has been edited by netfan: May 10 2006, 06:25 AM

Well the catch now is u have to pay 10 euros

well anyway I dont think there r any interconnect between all 015 ISP .... so u can call ur own cloud only ...

I ve tried using usb phone and other ATA or IAD ... well frankly speaking if i am using softphone also there are seldom echo or jitter or noise ... also calling to pstn ... if i am using TMNet 015

well try 1st la ...

This is an interesting topic of discussion. hehe :-)
Here are some issues id like to add:
Who is the target market for VOIP in M'sia?
if streamyx VOIP its running thru broadband, why should I pay for another telephone line since I have a fixed line. if its for cheaper call rates overseas, how different is it from Skype?

the target i think is for the current streamyx user ...
well it is cheaper then the standard STD on a fix line .... and IDD ... well comparing to skype ....

Skype have 2 type ...

1.Skype for pc to pc is free

2. skypeout is calling to pstn & mobile it work on prepaid basis ...
Malaysia euro 0.020 x RM4.58 = RM0.09
Malaysia - Mobile euro 0.047 x RM4.58 = RM0.21

more or less like tmnet one just that tmnet one have monthly ....

I am using Jaring my015. Their rates are somewhat cheaper then TMVOIP Service . I can plug any phone into it , as it uses the standard RJ-11 port. Its cheaper to place a call to Singapore , USA , etc.....

As for me , I am using Apple's iChat VOICE Chat service , its pretty good as well....

015 is not consider a fixed line.

Well... the 10 euros will last you for 120 days and if you make a lot of calls to Malaysian PSTN lines...

Calling to Singapore Mobile is also FREE!!!

Another VOIP provider you might consider is Astratel from Australia. Their mobile calls to Malaysia are about 15 sen and to Fixed line about 3.3 sen.

Just my opinion...

Prasys...

Just wondering how the quality of your Jaring my015 is? I was using mine to call mostly mobile numbers but the latency was high and also a lot of echo.

Would you mine sharing your settings with us? Are you using the Linksys PAP2 from Jaring?

well we r free to try any service right

I havent try jaring my015 or NationCall o15 one yet & dunno what r the latency & echo for both ... the one that i have tried is the TMNet 015 one .... i can say it is quite good when calling STD & Mobile using a streamyx 512 or 1M ... no echo or latency ...

as for high latency or echo problem most prob coz or high bandwidth utilization ... too much of seeding & leaching maybe ???


well u can consider that 015 is not a fix line ... some like a virtual line ... u can only call out to pstn or mobile ... not vice versa .... but 015 to 015 can la ...

uh .. I just heard that there is 1 vendor who is selling an USB thumbdrive which u can install ur softphone inside it and it will run automatically .... no need to install the sw in ur pc ... so very good if u r on the run all over the world ...
u only need a pc with a dsl line

more info:
IOCell Info

the vendor in malaysia hp
Delltone

cherioooo

Using MoBif VOIP. Fully 'Made in Malaysia' version of Skype and has the ATA (hardware version) made by the company itself.

how is the rate eh ... for STD & Mobile ?
the ATA look fimiliar .... ehmmm ...

how is the quality???

you guys should try voipstunt! Download the software from www.voipstunt.com


just pay using your credit card RM40-60

There is a VOIP software that can help us save lots of money. Try to download this program, you can call for free to various countries around the world totally free (only if u buy their credit) Credit expire around 5-6 months.



malaysia (Landline) free to call
australia (Landline) free to call
canada (Landline & Mobile) free to call
denmark (Landline) free to call
france (Landline) free to call
hong kong (Landline & Mobile) free to call
new zealand (Landline) free to call
singapore (Landline & Mobile) free to call
south korea (Landline) free to call
taiwan (Landline) free to call
united kingdom (Landline) free to call
united states (Landline & Mobile) free to call
lots lots more!!!!



The URL as below :
http://www.voipstunt.com
http://www.voipstunt.com/en/rates.html

The call quality = very satisfied!


My intention of this post is not promoting this product. I want to help you guys to save more money on the phone bills.

This post has been edited by ikuwara: May 11 2006, 11:36 AM

yup.. also use voipstunt.. sharing with my brothers n his friends.. but only 1 person can make a call at one time..

I use with ATA device connect to analog phone..

how much is the registration for TMNET VOIP??

voip stunt is not bad......you can try to register an account and try to make a call, you get to speak for 1min for free. try it before you buy

Could we make this a sticky topic? I am quite new to VOIP and still trying my best to configure my hardware.

I still think voipstunt or internetcalls is the best because you can make free calls to a lot of locations.

Hi guys,

Actually, i have been using VoIP for around 1 year with Linksys PAP2. The service provider that i am using is iptelnow (www.iptelnow.com). The rate is quite cheap as i call to australia in regular basis. Free call between extensions (in same network). AUD0.10 per call to australia landline.

I felt malaysia voip provider is quite rip off and cheat our money as their market is monopolized. they are the winner all the time hence they dare to collect service/subscription fees for nothing.

iptelnow that i am using now, i pay absolutely nothing if i don't make any call to PSTN. no minimum maintenance fee which i think is good. As i understood, VoIP calls quality doesn't depend on the service provider but our internet connection.

You can use a UK based voip service provider but calling to your friend in malaysia from malaysia with same quality that u uses TMnet or jaring voip service. because once the connection is establish, mainly is more to like peer to peer connection.

just my opinion. cheers.

Great find! Looks like IpTelNow allows you to BYO device. Nice!

Have you try calling another IPTelNow user in Malaysia (either on Jaring or TMnet) or the PSTN line in Malaysia? How's the voice quality like?

I have a PAP2 too that I use with US based FWD provider, which allows me to call another FWD for free but unable to call landlines at the moment. I may want to try check out this IPTelNow too since their call out rates are very reasonable. :-)

This post has been edited by netfan: May 13 2006, 07:35 AM

Anyone uses the SPA3000 with Voipstunt? Totally confused on how to do the configuration Can anyone help?

Doltan

You might want to have a look at this website (registration is free):

http://voxilla.com/ataconfig.php

Voxilla ATA Configuration Wizard

This FREE service, open to all registered Voxilla users, allows you to easily configure analog terminal adapters and IP phones from four different device manufacturers with more than 20 different service providers. We currently have Wizards for the following devices:
Grandstream Budgetone/Handytone
Grandstream GXP-2000
Leadtek BVA805x
Linksys PAP2-NA
Linksys (Sipura) SPA Localization Settings
Linksys (Sipura) SPA-841
Linksys SPA-941
Linksys (Sipura) SPA-1000/1001/2000/2002/2100
Linksys (Sipura) SPA-3000
Linksys (Sipura) SPA-3000 with Asterisk

If you are a device manufacturer and would like us to include your device in our Configuration Wizards, please contact us.

If you are a service provider, support open device configurations, and would like to be included in this configuration wizard, or we are providing incorrect service configuration information for your service, please fill out this form and we will do our best to correct the situation.

Doltan,

where you got SPA3000 and how much? wow, I have been looking for that product almost a year I think.. Not getting from voxilla coz quite high price when including posting, almost 1/4 the price of SPA3000.

Anyway, I know SPA3000 quite confuse to configure coz got many option. Can't help you there coz will only getting mine this weekend when I brother bring back from US.

Anyway last year linksys distributor quote me for RM650+ for that thing..

try to go to this link
TMNet EVP

yup better make this a sticky one

Bought it from Singapore Sim Lim Square for abt S$160. It was bought directly from the distributor there.

Have been to the Voxilla site before, but haven't used that config wizard before. I can go and try that again.

Hi, if i've bought my own PAP2 device, how do i go about getting TMNet's VOIP service?

From what i've read about TMNet Evoice, you need to buy the device from their respective agents (which comes preconfigured and LINE1 LOCKED). Is there a way for me to get TMNet's VOIP on line 2 plus me registering for it as well?

Thanks....

You can always buy from Jaring (which I think does not have a contract).

Cost is RM299 (PAP2) + RM10 (Stamp duty) = RM309

Jaring also gives you RM70 free calls which you have to finish within 1 month or else its forfeited.

Yeah, i know that, thanks....but my problem is still if i ALREADY own a PAP2 device....how do i add on a local VOIP service provider? I cant call TMNet coz the 1300 number doenst work from overseas....

jaring's VOIP kinda suck, lags like hell as i've used the JWBB from home before to call overseas....really teruk..!

Muncho gracias...

Go to

https://tmbill.tm.net.my/SelfCare/Maintenan...areMainMenu.jsp

and subscribe from there.

Ya.. a few of my friends is using iptelnow together locally in KL with me as we always conference together and chat. Besides, i uses it to call oversea regularly (especially Australia and HK). I tried FWD before i switch to iptelnow, the quality is fairly similar. Besides, using iptelnow we can make inter-network call to other VoIP provider like FWD which is free as well. but required to check the prefix.

Basically, the quality of voice is mainly depend on both party broadband connection and stability of the VoIP gateway.

Cheers.

Skype now offers free calls to any landlines in US and Canada for free until the end of year. So, no need for use up any Skype Out credits for calls in US and Canada.

This post has been edited by netfan: May 20 2006, 12:11 AM

I'm also using the Voipstunt, and now it can be use to call by using web pages access, and the sound quality is prefect.

Actually not 'web pages access'. You meant the softphone, which is really a small app on your computer.

Nowadays, it is more convenient to get a TA (Telephone Adapter) like the linksys PAP2 or SP3K (Sipura 3000), which has FXO port to allow direct PSTN connection. This way, you don't have to need a PC to make telephone calls.

'web page access is a new feature not many of us aware.. I also just aware about it and it really nice feature... No need ATA or even computer.. It more like 'sms callback feature' when you send the sms to initialize the call..

This web page access is where your login to ur account.. Put ur phone no in 1st box and no you wanna call in 2nd box.. Then it will call ur no 1st after you pick-up it will call ur destination.. It will charge for 2 call but since call to landline is free so it both way FREE. Unless you call to country not cover by voipstunt.

It would be nice if I have a lot of people to call in US or Canada but as it is, I do not know a single person there.

I agree!!! I was not aware of it only until a few days ago. I haven't tried it but www.internetcalls.com from the same company also has it. Charges are quite high though if you call Malaysia mobile to mobile i.e. 8Euros which is about 38sen comparable to a prepaid call.

Wow!!! Didn't know that I was that convincing! Anyway, cheers to this new sticky thread and hopefully everyone contributes to make VOIP in Malaysia on par with other countries. Do share your experiences that you have read in other VOIP forums.

A good place to start reading on VOIP related articles would probably be

http://www.myvoipprovider.com

Cheers again and happy reading!

just register at tmbill.tm.net.my ... but i dont think u can register unless u have streamyx .... but actually u can set the ata to streamyx 015 ... wz the right setting

This post has been edited by cruzzmz: May 21 2006, 10:06 PM

Wow!!! Globe7 is offering rates to Malaysian numbers as low as 3.66sen fixed line and 15.6sen to mobile!

Please check this out at http://globe7.com/rateplan.php?code=M

Haven't tried it so don't know how good it is. Anybody who has tried, please let us know.

A Cool New Mobile VoIP Application PDF Print E-mail
MyVoipProvider, 21 May 2006

A small australian company (www.voxalot.com) is causing headaches for fixed line and mobile Telcos around the world with new innovative VoIP products enabling users to bypass expensive traditional phone services.

Only two months ago they launched a new Web activated VoIP application allowing users to sign up to almost any SIP based VoIP provider(s) and with the help of Voxalot's service connect any two phone lines in the world while paying only VoIP rates. Best of all this service is free of charge and does not require any additional equipment, headsets or software to download.

Voxalot have now taken their application a step further by launching a mobile version. As long as you have a GPRS enabled phone you can connect to their service and launch a call between any two phones (landline or mobile) in the world. This little application is especially of interest to anybody travelling or without direct access to the internet.

The service is still in an early test phase and it is not yet possible to change any configurations while browsing on your mobile phone, but developments are underway.

The idea is brilliant for those without access to broadband and the implications are far reaching:

1. Never pay ridiculous hotel phone bills again
2. Currently one only requires internet access to setup a list of contacts. From there onwards all one needs is a GPRS enabled mobile phone.
3. Some mobile operators in Europe and elsewhere are starting to ban VoIP calls over their network. Voxalot's service bypasses this "ban".

No doubt that this little application will develop further and cause waves in the industry. For further details please check out their site at www.voxalot.com.

VoIP use on the rise PDF Print E-mail

Techworld, 18 May 2006

By Tim Greene, Network World

Large companies are leading the way in adopting VoIP adoption, adding it to their corporate phone options as their traditional PBXs age.

That's according to a new survey from Infonetics Research, which found that 36 percent of large US corprations are already using VoIP equipment and VoIP services.

Only 23 percent of mid-size and 14 percent of small businesses have adopted VoIP gear and services but Infonetics said this percentage is set to triple by 2010.

The main reasons businesses give for deploying VoIP include integrating phone systems across different sites and converging voice and data networks. But they also say they hope to save money on the cost of operating their phone networks, in part by paying less for long-distance calls.

Infonetics has conducted a similar survey for the past three years and they show a steady rise in the use of V0IP that will continue for the next three years without a dramatic up-tick in the adoption rate, according to the author of the study, Mattias Machowinski.

He says the steady increase is due to businesses turning to VoIP as their traditional phone systems reach the end of their life or as voice service contracts expire. At the projected rate of adoption, a half to two-thirds of large businesses will use VoIP by 2010, he said.

Among the businesses surveyed, the most often used IP PBX vendors were Cisco, Avaya and Nortel, the study says.

The study compared the amount organisations spent on VOIP services in 2005 and compared that figure to projections of what they will spend in 2007. The amount spent on hosted VoIP services jumped from US$47,667 to $63,799; the amount spent on managed PBX services jumped from $10,865 to $28,367.

The study also finds that the percentage of users accessing VOIP via Wi-Fi grows from 5 percent in 2006 to 20 percent in 2008.

The study was the result of interviews with 240 businesses using VoIP now or that will by 2007, as well as the results of 450 shorter interviews to determine VoIP adoption rates.

Xybirium,

Thanks for posting some interesting news tidbits on VoIP providers.

The news clip about VoxAlot is interesting but in my opinion, their procedure for web-call back feature is not really ground-breaking or can cause big concerns for regular POTs providers for potentially losing revenue over their traditional phone calls. For PSTN-to-PSTN, you have to select two VOIP providers that can give you the best rate for untimed calls. Why the hassle? You still required to have a PC connected to the Internet, and if you have that, why bother, just use a softphone to make the call directly.

Besides, most VOIP already provided hefty discounted rate for IP network to PSTN network. Unless one makes a lot calls over the same number between two providers, web-call back option is not really attractive at all. Besides, their claims of "free" is misleading because of this disclaimer at the bottom says "This service is free for all VoXaLot members during our initial trial period. Once the trial completes, it will be available to Premium VoXaLot members."

Today, a lot (unfortunately, not in Malaysia) of traditional phone service carriers are already embracing VoIP as additional call services within and outside their service areas with their own ATAs (or BYO) and using their regular data network in their core backbone to transport voice calls over to the Internet gateway.

The fears that the incumbent carriers to ban VoIP is really unfounded. What we will see is the incumbent carriers won't ban VoIP but participate in this new business model with their choice of VoIP gear and market it directly to their customers plus value-add it with QoS over their network (meaning their infrastructure). The other VoIP providers that doesn't have any infrastructure may have tougher time to sell later because they can't provide the level of QoS as compare to the incumbent who can.

For example in Malaysia:
TMnet can sell their VoIP with better service quality than 'Ah Beng' VoIP service shop. Why? Because TMnet owns the infrastructure and they can control it.

This post has been edited by netfan: May 23 2006, 12:49 AM

netfan,

most tier-1 telcos in malaysia are already utilizing data network to some extent to transport voice calls as their backhaul link. i know my telco does and all those mobile telcos do. don't know about the big one. multiple E1 PRIs can be very expensive to support so VOIP is a huge cost advantage in terms of loading up calls on a single pipe.

This post has been edited by ihsan: May 23 2006, 01:22 AM

Try it out http://www.globe7.com/index.php?refcode=R1172212F
(If ok, please use the referal code above to help me to get more free minutes.)
Free 100 minutes per account registered.
Now I have 10 accounts.

This post has been edited by choonkeong: May 23 2006, 02:11 AM

netfan & ihsan

I agree with you. Even if we do subscribe to all those overseas VOIP providers, their call quality still leaves much to be desired. On the home front, even Jaring MY015 quality is still bad. If you make mostly voice calls, the TMNet Prepaid ONE option is still a very good choice as you can make calls from any fixed line (including payphone) at the low price of 18sen. Voice calls to other countries e.g. Australia is only 20sen which is the same as the rate from TM's 015.

In Malaysia, I suspect a lot of large corporations are using VOIP to call their customers and I would bet that their solutions are from TM. This is because when they (credit card companies) call to request for payment, the quality of the call is really bad.

Thanks again for participating in this thread and I suspect others would be very grateful if you could share your views and experience.

This post has been edited by Xybirium: May 23 2006, 08:24 PM

Already signed up about a month ago. Just wondering if you would like to share your credit. Hehe...

Hmm.. you mean like IP-based Contact Centers (aka Call Centers)?

Poor voice quality can be caused by anything on the network. It could be due to telco network congestion or it could caused by end-user equipment too.

If one has a converged network, and if there's problem, unfortunately, the blame always seems to hit the carrier first.

In order to provide good service quality (for both their data and voice network), both
the corp/enterprise and telco have to work together to build and put together a troubleshooting plan for repairs.

Most telcos should be able offer some form of QoS with SLA to differenciate packet-voice and raw data traffic within the boundary of its network (and also to isolate the troubleshooting domains). But, if the end-users POTS lines are provided by another carrier (like when it bridges off to a mobile phone carrier or another central office that has older equipment), it may be difficult to give service guarantee end to end.

This post has been edited by netfan: May 24 2006, 12:55 AM

most VOIP SLAs only covered areas which the provider can guarantee ie: if they own the network. in the case of dialing other provider numbers you can't certainly guarantee SLA of your own if you can't get just as stringent SLA from other providers which interconnect with your provider.

for example, how do you provide voice SLA when you know there will be a translation between VOIP->PSTN when the call crosses a transit/end provider which does not have VOIP enabled in their backhaul and it affects your SLA? to put it simply, you can't and you would be naive enough to believe that any providers can provide end-to-end SLA.

even IP-IP calls can also be affected if you have provider using MP3-quality voice calls and the end provider is using a lesser call quality ie: codec translation.

in short take it with a grain of salt if any provider can offer end-to-end SLA with regards to VOIP or even IP network. unless they own the network themselves, they are certainly skewing every means to get your business.

This post has been edited by ihsan: May 24 2006, 01:30 AM

netfan & ihsan

Just out of curiosity, where are you 2 guys/gals working anyway? You seem to be very, very knowledgable in VOIP.

recently i had this problem when trying to make conference call.. i can call 1 person no prob... when i conference an automated voice says " Your account is in use, thank you"

Whats going on ?

Ur account does not support more than one session of call..

I agree. Sometimes, the marketing dept of seems to overhype a product that they're selling. You have to really read the fine print with the magnify glass or have a lawyer to examine it.

Most telcos/carriers are only able to provide SLA for PE to PE within their cloud.

However, as I have found in a recent a RFP briefing with one carrier in US, they're are able to provide CE to CE SLAs and over another carrier networks with an established (private) NNI as well. Somehow, they're have stitched together an extended MPLS network over multi-carriers between two cities approx. 2400 km apart. Come to think of this, this can be difficult if both carriers' class of service model doesn't fit.

The question comes to mind is how does one give QoS priorization over another enterprise voice traffic if both are competitors (and futhermore over multi-carriers)?
eg. who do say my voice traffic is more important than the others? Huh?

Ihsan:
Btw, do you know if any NNI for MPLS is being done in Malaysia? For example: TMnet working with Jaring or maybe Maxis working with Digi on their IP network?

This post has been edited by netfan: May 25 2006, 07:48 AM

here where i work, we are certainly looking into it but there hasn't been any proactive movement towards it from the big ISPs unfortunately, since they believe the market is theirs. inter-provider IP-VPN is certainly a viable achievement, speaking from a technological standpoints but the only thing stand in between is just political/business egos.

xybrium,
i work with a local ISP but my specialization is not voice. i know a little or two on how phone works and that's about it.

This post has been edited by ihsan: May 25 2006, 08:15 AM

Sorry, can't reveal specific but I am working in research and educational network (not in Malaysia) with direct dealings with a lot of Tier1/Tier2 ISPs. However, I still considered myself a newbie in VoIP. Voice is not my specialty either. I am router and switches guy.

In the past, voice is solely the phone technician domain. For example, if the phones doesn't work, just call the phone tech at local telco, he'll come and to fix those bad copper lines or replace your phones.

With VoIP, if the calls have problem, the IP phones or the IP-PBX doesn't work, now, the network sys. admin or IT guy has to get involved. Heck, the noisy user will ask, hey did you bring down my network? Did you prioritize my traffic? blah-blah..
Somehow the network always get blamed.

Another thing, 'Triple Plays' ads hype are begining play on those CEO's mind. Sooner or later, we'll see tech support calls like 'TV doesn't work' - call your network administrator (whatever happen to call the local TV repair man?).

Hmmm, come to think of this, I don't think I like where it is heading.

This post has been edited by netfan: May 25 2006, 08:23 AM

What is Fujitel ?

Its a service allows home users, businesses and even larger enterprises to talk to each other seamlessly, and for a lot less than you would expect. All you pay is an Internet device and your monthly broadband service while the rest is FREE !


Key Benefits of Fujitel VOIP

High quality telephone technology

Turn long distance calls become local call charges

Drastically reduce call costs and reduce line rentals

World wide on-net calls for Free

Turn your PC into a telephone

Country RM/min
Australia 0.10
Canada 0.10
China 0.10
France 0.10
Germany 0.10
Hong Kong 0.10
Italy 0.10
Malaysia 0.10
Norway 0.11
Singapore 0.10
Spain 0.10
Sweden 0.10
Switzerland 0.11
Taiwan 0.10
United Kingdom 0.10
USA 0.10

Contact: Mr. Yong-017-3831879
: : :

I'm sorry if I rattled you but you just seemed to be experienced in these. However, as I said before, me and all the rest of the thread readers will gain very much from your experience even if you are just a switches and routers guy.

Thanks again.

I think this is supposed to be in the garage sales thread and not here!!!

Not a problem at all.

VoIP is a not a new thing for those already familiar with the video conferencing (using H.323) environment. It is a simple transition over to packetized voice network. The data network guy now has to learn and understand the old traditional circuit-switching PSTN to make it all work together.

With so many improvement has been done with SIP protocol (for call setup, directory, etc) today, many manufacturers making their ATAs, IP phones, IP-PBX, all have SIP capabilities, making it fairly easy to interface with one another with different vendors.

Having the global IP network (ie.Internet) all stitched all system together, it only make sense to use this network to transport voice as well (yeah, we all know it is best effort esp for those need to call overseas).

There are many concerns about VoIP, particularly in Security and Emergency Services.

Security: The IT network guys know this. The malwares/viruses that spread to your PC, this can happen to your IP-phones,IP-Based PBX. Your VoIP network is subject to same DDoS attacks that plague the IP network. Without proper planning, the voice network will go down with it. Other disaster issue is: Power outage. Traditional phone can still hear dial-tone. The IP-phones is dead if you don't implement PoE (Power over Ethernet) with PSTN trunk line backup.

Emergency Services: These are good questions to ask those newbie VoIP providers.
What does one (say a regular user) expect, when he/she picks up the phone and dial '999' in an emergency situation? Does emergency service (eg.police) know where you're calling from? Is your VoIP provider aware they're responsible to setup a call service to route your calls with all your info to the emergency service provider? Those are important things to consider before we migrate away from our traditional PSTN service.

This post has been edited by netfan: May 27 2006, 12:42 AM

hi guys, its good to have this thread. Many people/company penetrate on this VOIP recently for sving their bills.

i also involve a little since it can make money (if you really2 in the business). for now i'm searching for a good ATA that not too expensive for market. i guess china got a lot.

abbybody using ATA at home?

I am using Aztech 600EW ADSL Modem/Router and Linksys PAP2 ATA but I have not been able to configure it properly to work so far.

Previously I could make calls but it was scratchy and latency was quite bad plus echo. Now I don't even hear a ringing tone if I call the other party (e.g. my own handphone rings when I call through VOIP but I can't hear any tone using my ATA). If there is anybody who can help or point me to the proper website, I would be eternally grateful.

Try setting up with free FWD or VoIPStunt accounts and call yourself using Phone to PC. If this works with no problem, then it must be your VOIP provider.

Anyway, if you're paying your VoIP service with the supported ATA device, your provider should be able help diagnose it as well. PM me if needing more help.

This post has been edited by netfan: May 27 2006, 12:42 AM

Can I use VOIP with an EDGE internet connection?

when i was in asp,(autodialer system) that sort of things always happen. i was the technical dpartment. i hate hearing customer complaining about echo, no ringing tone when connected, no ringing on the phone, bla bla bla sometimes misdirection to air tawar and other places.

i check the dialers, its fine. then check with my network dpartment, the billing record, bla bla and the answer fine too. then we go to our termination partner, we found the faulty on them. sometime telekom also makes problem( they hide the issue coz worry we want to claim for rebet.)

I guess its the same with ata. the ata wont make any problem coz it doing its job detecting dtmf, transmitting bla bla. the termination partner also the same.but we using ISP as our lines for calls. i think the problem is there but i dont know how to config or do somthig to make it right.

correct me if im wrong

got very good thread inside here ...

well voip is a good way to make call cheap ... (that is what we were told by the marketing side) actually it is the codec that make the most of VoIP sounding much more better than the old POTS phone ... but this is depending on how the network is set up ... if u have MPLS and u priotise the traffic to be voip 1st then maybe u will get a better QoS ... but that can only be done if you have the control all over the network including ur cpe (which have a capibility of ctrling ur trafic) then ur Edge routers ... then for ur core router ... ISP have to offer a SLA of maybe a standard & premium class of package that will cost more but can have a good SLA between ISP to ISPs ... so then we can ensure all is good along the way.

As for lantency... well to troubleshoot you have to see your lantency if the roundtrip is more than 300ms then it is very diff to use VoIP ... i am not sure what is the lantency with Jaring015 ... well as for calling pc to pstn the problem there is we cannot control the 2 network of IP and PSTN cloud ...

well that is only my opinion



This post has been edited by cruzzmz: May 27 2006, 08:46 AM

I am actually using VoIPStunt but haven't tried free FWD yet. Previously, it worked fine. My mother used it to call my aunty in Australia and when I enquired about the quality, she said it was fine and didn't detect any difference from the normal POTS (and furthermore its free!!!)

As you said it may be because of the VoIP service provider or it could be my settings. In the meantime, I will try to troubleshoot it myself because for me, it would be a learning process. I'll PM you if I'm really desperate. Thanks again.

Thank you for sharing your experience with us. Keep it coming as we need everyone involved.

By the way, where can I read more about autodialers? I'm still sketchy on what it is and how it really works? e.g. can I attach it to my fixed line and how much it costs. Any feedback would be very much appreciated.

Maybe yes! With the right mobile phone and the right software. Lookup on Skype. I think they are trying to get more people to sign up (FREE)!

If you find any interesting websites, you can post it here so that all of us can learn something new.

It really is good to see more people involved in this thread. I was also using Jaring015 previously. The latency I usually get on average is about 50 - 100ms. Same for TMNet.

I haven't been able to set my QoS settings in my router properly as yet as they don't seem to be good guidelines on that. I even printed the manual for Aztech 600EW and after reading through it, still could not understand it properly.

Guyz, I don't know if you you noticed the VoipStunt has a hidden clause:-
"Max 300 minutes per week of free calls, measured over the last 7 days. Unused free minutes cannot be taken to the following week(s). If limit is exceeded, a minimal rate of 1ct/min will be charged (as shown in the VoipStunt software)."
Directly below the website..
What do you guyz think? Is it possible we'd be charged somehow?

Yup, I noticed it a few days ago. Must be their service is getting so popular that they have to do that. However, if you always make lots of calls to fixed lines, even in Malaysia, I still think it's worth it. Where in the world can you get to make "free calls" of 4800 minutes at only 10Euros or 0.002Euros or approximately 1sen per minute?

Oh... by the way, iTalk Mobile is offering 15sen/minute call to Malaysian Mobile numbers from fixed line (business/home line & payphone) and 25sen/minute from mobile line (using missed call back).

I'm just wondering if any other companies in Malaysia can beat that, but then again, it's TM.

anyone know why isit so ? before this i can make conference call up to 4 people... whats wrong uh ? anyone get this problem ?

Autodialer is a past now, i think you dont hv to look 4 it anymore coz it not giving saving or lower rates as far as VOIP. What happen to company that using autodialer now? they will be push to use totally VOIP or they will be left behind. (my english is suck. haha)

A picture of autodialer on single line. meaning ur home fixed line.


A picture of branches using ATA from VOIP Provider for calls, bla bla bla...


i'm still newbie for voip, dont even hv account for 015.. u guys here already own it and make conference call.

i'll use what u guys recommend. then i wanna buy an ata for trial and business. its worth invest in VOIP

This post has been edited by thenobodies: May 28 2006, 02:26 PM

anyone want to use VOIP for one day.... unlimited calls... call anywhere... how long u want... RM 100 for 24 hours.

Well this is e-voice ( my personal account), what u do is, bank in RM 100 to my account and I will give you the username and password for you to start calling... after 24 hours ill change the password. Simple... hehe... anyone want to try PM me or email me rattan77(at)gmail.com

which provider u r using ??? maybe they block the conf cos of missused (hint)

This post has been edited by cruzzmz: May 29 2006, 09:30 AM

well as for setting up QoS at ur ATA it will not be any diff it ur all the way in the cloud there is no QoS set ... well we will have to wait until there is a proper SLAs for the cloud ...

as for the round trip i can see that mine ... if i ping to TMNet 015 it will only return average of 6ms .... ehmmm and I am using wireless
but then again I will test it on my streamyx line ....

cheers

cuzzmz : im using e-voice premium (TMNET)

this is what i get after running a trace route to evp.tm.net.my

Tracing route to evp.tm.net.my [219.93.150.9]
over a maximum of 30 hops:

1 1 ms <1 ms <1 ms 192.168.0.1
2 51 ms 45 ms 44 ms 219.93.218.177
3 49 ms 43 ms 43 ms 219.93.217.197
4 50 ms 42 ms 43 ms 210.187.132.49
5 45 ms 46 ms 45 ms brf-odsy02-srp1-0.tm.net.my [210.187.135.2]
6 379 ms 46 ms 44 ms 210.187.143.1
7 47 ms 148 ms 44 ms 219.93.182.230
8 51 ms 44 ms 44 ms evp.tm.net.my [219.93.150.9]

Trace complete.

mine ....

Pinging evp.tm.net.my [219.93.150.9] with 32 bytes of data

Reply from 219.93.150.9: bytes=32 time=11ms TTL=59
Reply from 219.93.150.9: bytes=32 time=10ms TTL=59
Reply from 219.93.150.9: bytes=32 time=11ms TTL=59
Reply from 219.93.150.9: bytes=32 time=10ms TTL=59
Reply from 219.93.150.9: bytes=32 time=9ms TTL=59

Ping statistics for 219.93.150.9:
Packets: Sent = 20, Received = 20, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 9ms, Maximum = 14ms, Average = 10ms

Ave 10 ms quite good

trace route

Tracing route to evp.tm.net.my [219.93.150.9]
over a maximum of 30 hops:

1 * * * Request timed out.
2 17 ms 17 ms 17 ms 219.93.218.177
3 28 ms 20 ms 9 ms 219.93.216.189
4 10 ms 9 ms 9 ms 210.187.142.1
5 10 ms 11 ms 9 ms 219.93.151.102
6 10 ms 11 ms 9 ms evp.tm.net.my [219.93.150.9]

Trace complete. ...

I use a aztech ATA300 .... well i dunno why cant make call conf to pstn .... maybe they dont allowed it anymore ... but conf onnet to onnet still can i hope

Pinging evp.tm.net.my [219.93.150.9] with 32 bytes of data:

Reply from 219.93.150.9: bytes=32 time=9ms TTL=59
Reply from 219.93.150.9: bytes=32 time=10ms TTL=59
Reply from 219.93.150.9: bytes=32 time=9ms TTL=59
Reply from 219.93.150.9: bytes=32 time=9ms TTL=59

Ping statistics for 219.93.150.9:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 9ms, Maximum = 10ms, Average = 9ms

Nice pings. This ping test was done when my CS server was running too.

try using ethereal to sniff up ur conversation and maybe can post the jitter result here

Target Name: evp.tm.net.my
IP: 219.93.150.9
Date/Time: 29/05/2006 07:13:52 pm

2 * * * * * * * * * * [-]
3 * 24 ms 59 ms 24 ms * 23 ms * * 24 ms 23 ms [219.93.218.117]
4 24 ms 23 ms 23 ms 23 ms 23 ms 23 ms 23 ms 23 ms 23 ms 23 ms kbu-odsy01-g0-1.tm.net.my [210.187.133.241]
5 58 ms 49 ms 139 ms 78 ms 228 ms 49 ms 48 ms 49 ms 133 ms 48 ms [219.93.172.117]
6 87 ms 48 ms 49 ms 50 ms 49 ms 48 ms 48 ms 49 ms 55 ms 50 ms [210.187.126.218]
7 49 ms 49 ms 50 ms 49 ms 49 ms 48 ms 49 ms 49 ms 49 ms 57 ms [203.106.206.54]
8 47 ms 46 ms 46 ms 46 ms 46 ms 47 ms 46 ms 46 ms 47 ms 46 ms brf-odsy01-srp1-0.tm.net.my [210.187.135.1]
9 46 ms 47 ms 46 ms 222 ms 46 ms 46 ms 47 ms 47 ms 47 ms 46 ms [210.187.142.1]
10 50 ms 50 ms 50 ms 49 ms 50 ms 49 ms 49 ms 50 ms 50 ms 49 ms [219.93.151.102]
11 46 ms 46 ms 46 ms 45 ms 45 ms 45 ms 46 ms 45 ms 46 ms 45 ms evp.tm.net.my [219.93.150.9]


This post has been edited by Xybirium: May 29 2006, 07:15 PM

hey cruzzmz how does ethereal work ? mind to explain since u know about it ?
could not understand much from here :

http://www.ethereal.com/introduction.html

well i usually use a few sniffer but ethereal is free plus can sniff VoIP packet ... u just download & install on ur pc ... but i think u still need to install
winPcap
then u can sniff by using a softphone on ur pc ....


cheers

This post has been edited by cruzzmz: May 30 2006, 11:00 AM

thenobodies:

Your examples of "autodialers" use is mainly for telemarketing business, isn't it?

The voip for business basically may save you on the long distance toll-call charges but telco providers are smarter than this. VoIP for business rate is not the same as the residential service rate. Their rates are most likely to be higher.

Whether it is VoIP or Traditional Circuit Switching PSTN, you still need that "autodialer" mechanism to make that volume calling behind the scene. Making it to work may cost you more.

Be warned, some businesses have actually lost more customers after they have moved to VoIP-PSTN for calling because of poor quality, loss calls and frustating call agents. This is not because of the technology, it is because of poor implementation.

The low cost best effort type of residential broadband type of connection is probably not good enough to provide quality and stable network for running a business. You may have to go for dedicated leased line.

This post has been edited by netfan: May 31 2006, 06:30 AM

Here's something to share: VoIP in Osceola School, Florida, USA

A video clip of a success story of VoIP implementation in a US public school by Cisco Systems and a local solution provider - Data Dimension.

Hmm..I wonder if there's any public school in Malaysia has VoIP?

This post has been edited by netfan: May 31 2006, 06:18 AM

I agree with that .... it is better to have a dedicated line for ur own VoIP ... regardless it is a DSL line or better leased line ... the codec also plays an important role to the bandwidth consumption .... plus the QoS

As for the malaysia school that implement VoIP not sure who got it ... but maybe smart school is going to do it ???

I not sure whether all smart school got VOIP but I know some that fell under schoolnet projet got VOIP as part as internet services. Those school got VOIP which allow them to call each other free of charge.