On 31th of March 2008 our GTalk2VoIP gatewaying service has adopted a new, and we suppose, very powerful feature: now any VoIM user (I'm bit tired typing in all the messenger names) can buy a real phone number within any of a large list of coutry codes (at the moment 40+ country codes are availabel) right from their personal account page (MYPAGE) and start receiving phone calls right into IM messenger. The calling service is free of charge, but one has to pay monthly and setup fees for the phone numers he or she is subscribing to. Fees vary from $2 a month depending on country code and phone number vendor. The cheapest numbers are available in US/North America country code - $2/month for a phone number in +1-567 area.

Every phone number purchase is provided with 12 hours of testing period during which user can release recently bought phone number and get 100% refund (money back). So we encourage every one to try this new service, even if you won't need it or find it not of sufficient quality you can release your numbers withing 12 hours and get you money returned to your balance. We also want to hear your feedbacks on this service and a list of country codes you think that we are missing.

Details are here http://www.gtalk2voip.com/gtalk_service_did.shtml

Regards,
Ruslan.

Talkonaut for Symbian S60, pre-release.

For the last few months we have been working hard on preparing for release of our new and native Talkonaut mobile software for Symbian S60 platform. Although we devoted a lot of time tesing it extensively, there is always a room for a bug. So we thought that you might be interested in helping us with testing process. You may download a pre-release version of new Talkonaut from http://www.gtalk2voip.com/~rz/talkonaut-s60-3rd.SIS. We expect to hear constructive feedbacks from you, so if you are not ready for "undesirable" features, you better not try it at all :).

Meanwhile, I managed to arrange some time to write a small essay regarding our mobile voip development process and all the experience we got. Here is an excerp:

Since the very start of GTalk2VoIP project early in 2006 we at GTalk2VoIP TEAM had been nursing the idea of creating something very Google Talk like for mobile phones which could be used to make voice calls over GPRS (and WIFI) as simple as it is made in Google Talk: you see a contact of your friend, if he/she is present, you press Call button and you talk. So easy and so powerful, isn’t it ? On the opposite, the mainstream consumer VoIP software (so called soft-phones) was and still is very unfriendly: simply downloading a SIP phone onto your PC won’t make you able to make calls over the internet instantly, you will need to obtain a SIP account, configure it into your soft-phone, define a dozen of other settings like STUN server, dialing schemes depending on your SIP provider, etc, and only after that, if you were lucky enough, you will be able to add your friend’s SIP URI into list of contacts, then call. Of course being able to manage as many settings is very positive feature for a power user, but for an average this kind of mainstream VoIP is very frightening and totally unacceptable. Sure, there are some projects like Gizmo Project (now Gizmo5) which are desperately trying to make things better, but still…

So at that time early in spring of 2006 we came to a thought -- we want all that Google Talk style VoIP on our brand new mobile phones, with all the power of presence and Jabber based IM chat, we want it to be able to run through dominating GPRS/EDGE service of GSM networks as well as through WIFI hotspots, we want it to be easily and automatically switching between available hotspots like any GSM phone switches between towers, we wish it had a Google Talk like user interface. Why ? Because it’s clean, neat and intuitive, there’s nothing obsolete in its design like innumerous features of Yahoo! Messenger, nothing irritating and slowing down like in Live Messenger, and it is not using stone-age protocol like AIM. But most of all we want it to be completely based on open standards...


Everyone is welcome to read the full text from http://www.gtalk2voip.com/why-i-hate-mobile-industry.html. Please, let me know what you think on the subject.

Regards,

Ruslan.

iChat support improved.

All Macintosh users are welcome to try our new feature of GTalk2VoIP gateway: now it supports voice callings for iChat users. Find technical details on http://www.gtalk2voip.com/gtalk_service_aim.shtml

Regards,
Ruslan.

GTalk2VoIP server upgrade

For the last few weeks our small team has been working hard and has successfully performed moving GTalk2VoIP service to a new server platform. The new server is lots more powerful SMP machine which is now located in Mountain View, CA, USA and has alot faster internet connectivity. We expect to see significant increase in performance and service stability, as well as latency reduce resulting in higher voice quality for all calls made through our gtalk2voip service.

New server will also allow us to introduce a number of new services. The most recent service we are planning to run into production is AIM/ICQ/iChat voice support. Expect to see it soon!

Regards,
Ruslan.

Powerful CALLBACK to set up multi-channel voice conferences.

IM initiated callback is now possible with GTalk2VoIP. You can connect any number of the phones, Google Talk, MSN, Yahoo or SIP users into a voice conference by issuing a single IMmessage.

All users of Jabber based IM chat, Google Talk, MSN/Live Messenger or Yahoo! Messenger can use GTalk2VoIP service to initiate VoIP calls using CALLBACK technique. This means, our system can make VoIP call to your phone (mobile or landline), then make a call to your destination and merge two calls (legs). Callback is initiated by a single IM message sent to service@gtalk2voip.com.

It is possible to add more participants into such callback organized calls thus creating a multi-channel voice conferencing calls. We encourage Web 2.0 developers to adopt this technique into their Web services and letting people freely and easily participate in voice chats.

To initiate a callback call, please follow these simple steps:

Step 1. Subscribe to the service in one of the possible ways:
1st Way. Open your GoogleTalk (or any other IM messenger) and invite new recipient whose user id is service@gtalk2voip.com.
2nd Way. Go to the main page of this site and submit your user id by pressing "Invite" button, then accept invitation from service@gtalk2voip.com.

Step 2. Open a chat window to service@gtalk2voip.com and send it a CALLBACK command using the following syntax:

CALLBACK [phone:gtalk:yahoo:sip:]SourceLeg [phone:gtalk:yahoo:sip:]DestinationLeg [via provider1 provider2 ]

Where, SourceLeg - is your own location, DestinationLeg - is a destination you are calling to, provider1 - is a provider name which will be used to deliver call to SourceLeg, provider2 - is a provider name that is to be used to call to DestinationLeg. If providers are ommited, then the system will implicate the default behaviour to call to each leg, i.e. it will call the best rated provider first (best ASR and ACD), if that fails, it wll call the second rated, and so on.

Each leg can be one of: phone number, Google Talk, MSN, Yahoo or SIP URI. You can use the follow syntax for each of the legs:

• +XXXXXXXXX or phone:XXXXXXXXX where XXXXXXXXX is a phone number (mobile or landline) in international format,
  
• gtalk:user@gmail.com where user@gmail.com is some Google Talk user identifier,
  
• msn:user@hotmail.com where user@hotmail.com is some MSN/Live Messenger user identifier,
  
• yahoo:user@yahoo.com where user@yahoo.com is some Yahoo! Messenger user identifier,
  
• sip:user@provider.com where user@provider.com is some SIP URI.

Read more details on http://www.gtalk2voip.com/gtalk_service_callback.shtml

Regards,
Ruslan.

Follow-Me feature for voice capable IM clients.

We now have Follow-Me feature. It allows VoIM users to receive incoming SIP calls and forward them to any mobile or landline phone or to any SIP URI. Usage is simple, you just define a phone number you wish calls to be forwarded to. After that, if you do not unswer incoming SIP call within 20 secs it gets forwarded to your Follow-Me number automatically. Same applys if you are off-line. If Follow-Me is not defined, calls are forwarded to Voicemail system.

Read more on http://www.gtalk2voip.com/gtalk_service_followme.shtml

Regards,
Ruslan.

GTalk2VoIP TEAM.

User-defined SIP accounts to make outgoing calls.
User-defined dialing plan.

As usual we spent lots of time coding and evaluating new ideas on our GTalk2VoIP gateway.
This time I would like to introduce another conceptually new and powerful feature: user-defined SIP providers. It's known that SIP services are becoming more and more popular on the net, most of them are kind of identical, but some are very different. For example, Gizmo Project lets you make totally free PSTN (phone) calls to almost any phone number of your friends listed in your roster; SIPNET let's you make free calls to largest cities in Russia; FreeWorldDialup lets you call thousands of other SIP users. So, we thought why not to gather all the power of SIP into one place and bring it to all VoIM users ? Why not to let VoIM users call to their friends' mobile phones through Gizmo Project for free ? For the last couple of months we were toying with this idea, we were thinking on how useful could it be and how to make it easily accessable for anyone in IM world. So, here it is. In brief, if you are a Google Talk, MSN or Yahoo messenger user, you can:

1. Define one or more of your existing SIP accounts (login, password, SIP server) in our system. You probably have SIP account and have been using them with a hard or softphone ? Now you can use them with your VoIM messanger. When defined, these accounts will be displayed only to you as alternative routes/VoIP providers (when entering CALL or COST command).

2. Define your dialing plan to route some destinations to cirtain SIP providers. You can use yours and other VoIP providers (registered in our system) together to define routing scheme appropriate for your needs.

You can find technical details on this feature at http://www.gtalk2voip.com/gtalk_service_custom_defined_sip.shtml

We hope you really enjoy this new feature of our gateway. If you have any questions or issues, just drop us an email to team@gtalk2voip.com. Thank you.

Regards,
Ruslan.