Tag Archives: VoIP

Google Voice — a personal Virtual Phone System

After waiting a couple days after applying, I received an eMail invitation for Google Voice. This service, formerly called GrandCentral before it was acquired by Google, allows you have a phone number which is independent of your actual phones. It allows you to create a phone number in an area code of your choice. Calls from this number are forwarded to any or all of your phone lines (home, work, or cell).

Domestic Long-Distance Calling

Using Google Voice, I’ll finally be able to ditch MCI long-distance service on my work landline. Calls within the U.S. and Canada are free. The way this works is that you enter the calling number from your list of contacts on the web site, and designate which phone (work, home, mobile) that you want to talk from. Google Voice will then ring your phone. Once you pick up you’ll immediately hear a dial-tone as it attempts to ring the called number.

International Calling

Calls from the U.S. to Germany landlines and Hong-Kong are 2 cents per minute. Some African countries (Gambia) are 33 cents per minute. Rates for to mobile numbers can be much higher; German calls are 18 cents when the recipient is a mobile phone.

Voice Mail

Google Voice includes a full-fledged voicemail service. There is a text to speech service which attempts to render messages into text, and then display them in an eMail-like list. I tried a similar service that was provided with a standard TrixBox installation.
The standard greeting isn’t anything like the graceful “Alison” provided with TrixBox and Asterisk installs; instead it something like, um, “yenta-in-a-hurry”. I’ll have to redo my own voice greeting shortly.

SMS Messaging

You can send SMS messages to mobile phones by typing the message into a web page. Great for those of us who don’t text.

Call Screening

You can screen all unknown callers (i.e those with out an entry in the contact list which include their caller ID), or you can screen blocked callers.

Notifications

Notifications of new voicemails can be sent to an eMail address, and/or mobile phone via text message.

Of course some of the usual things are also included, such as the ability to listen to voicemail from your phone, variable greeting by caller, the ability to forward voicemail. You can also record calls and store them online and create conference calls

So, Google Voice has definitely passed the Five Minute Test…. and it looks quite promising.

Call Centers from Hell and Customer Contempt

Especially during a recession, it amazes me the utter contempt company call centers show toward their customers. I spend a lot of time on calls with technical support people, and it remains as irritating as ever to get to them. Once I get a live person, however, I can usually calm down.

I really hate hearing that “This call may be monitored or recorded for training or quality control purposes”. Especially right at the outset of a call. Maybe for my broker… (what broker?) when giving financial instructions. For all the “training” that is going on sitting in phone tree hell doesn’t seem to be getting any easier. And quality control purposes? It makes me uneasy that I’m being recorded at all.

On hold, I really hate hearing every 15 seconds that “We appreciate your patience, and thank you for waiting during this brief delay”. and “We know that you are very busy, we appreciate your call”. Or worse, blabbing on about the web site, or the new product, or alternate ways to contact us, or whatever. These constant interruptions makes it impossible to concentrate on other work while waiting. What ever happened to playing Vivaldi and not inserting commercial messages?

While there are “secret” phone numbers floating around the internet for various services, I can’t imagine a company would want these internal numbers published; they would get spammed quickly.

So, I’m holding for Fairpoint right now, and have had 5 dumb “Thank you for holding messages” in the previous minute.
“Thank you for holding, your call will be answered in just a moment”
“We know your time is important, and appreciate your patience while on hold”
“Every effort is being made to ensure that your wait is as short as possible… Thank you”
“Thank you for holding, someone will be right with you.”
“Your call is very important to us. Thank you for waiting and bearing with us during this brief delay”.

And then the cycle starts again. All this accompanied by ear-splitting muzak (tacky fake FM-synthesized saxophones.)

Voice Over IP Updates

I haven’t focused on VoIP for awhile… but others do.

FierceVoIP announced that Logitech has bought out SightSpeed. Sightspeed was (is) one of my favorite videoconferencing applications, and it will be very interesting to see what becomes of the product.

VoipInsider reports that Polycom has updated the firmware for Soundpoint phones.

VoicePulse has announced a fail-over option for their accounts. They’ve completely redesigned their web site with a new interface, that looks really classy. This is one VoIP provider who appears to be here to stay. Hooray!

Windows for Devices reports that Motorola will discontinue development of Symbian and MotoMax phones, and concentrate on Windows Mobile, and Google Android. The site emphasizes hardware running embedded versions of Windows, there is a companion site for the Linux crowd at Linux For Devices.

Asterisk Alternatives?

In addition to Trixbox there is a non-derivative project called Freeswitch which is not based directly on the Asterisk open source code.  From the introductory web page:

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16 or 32 kilohertz and can bridge channels of different rates.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.