Disruptive Telephony

This turns out to be ridiculously easy to do in a Skype group chat. An administrator for the chat simply has to type in the chat window:

/set options +HISTORY_DISCLOSED

After this, anyone joining the chat will see recent history. Per Skype's FAQ on chat commands:

Joiners can see the conversation that took place before they joined. The limit that they can see is either 400 messages or two weeks of time, depending on which is reached first.

I've enabled this setting on a number of chats for which I am an admin, and it's definitely helped newcomers come up to speed on what is being discussed in the chat. (Of course, some of those chats are very busy and so 400 messages may only take you back a very short period of time.)

Note, again, that you must be an administrator of a Skype chat for this command to actually execute. You can type it if you are just a user in the chat, and Skype won't tell you that it didn't execute... but it won't. You have to be an admin.

BROKEN IN SKYPE 5.1?

Now, having said all this, I recently had two people join a Skype chat and not get any history upon joining. In asking what Skype version they were using, it turned out both were running the new Skype 5.1 on Windows. Did this chat history feature get broken in Skype 5.1? I don't know... and given that I have no Windows machines around to test, I can't tell you for sure.... but I thought I'd mention it in case you have people joining a chat and not getting any history. You may want to find out what version of Skype they are using.

P.S. Note that this is VERY different from the "/history" command in the IRC-style commands for Skype I recently wrote about. The "/history" command loads the chat history ON YOUR COMPUTER into the window for the chat. However, this history is only available for the time that you have been in the chat. You are NOT able to get the history of the chat before you joined. The only way to get that previous history automatically is if an admin set the option described in this post before you joined the chat. Once you are in a chat, the only real way to get chat history is to ask someone else to copy/paste their history either directly to you or in the chat itself.

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/help

This will work on old and new versions of Skype and will give you a list of what commands are available:

Available commands:
/add [skypename]
/help
/topic [new topic]
/me
/history
/leave
/goadmin
/get creator
/get role
/whois [skypename]
/setrole [skypename] MASTER|HELPER|USER|LISTENER
/kick [skypename]
/kickban [skypename]
/get guidelines
/set guidelines [text]
/get options [text]
/set options [[+|-]flag] ...
/setpassword [password] [password hint]
/clearpassword
/get password_hint
/set password_hint [text]
/set password [text]
/get banlist
/get allowlist
/set banlist [[+|-]mask] ...
/set allowlist [[+|-]mask] ...
/alertson [text to match]
/alertsoff
/call [skypename[,skypename,...]]

You simply type any of these commands into a chat and, assuming you have permission to run the command, the action will occur. The full list of commands and options is available at this URL:

http://www.skype.com/go/help.chathelp

If you are the administrator of a chat, you have access to the full list of commands. You are an admin if you either are the original creator or a chat - or were promoted to an admin by another admin using the /setrole command.

I personally use "/me" a good bit to attempt to add emotion into a text chat (ex. "/me laughs"). I also use /alertson and /alertsoff quite a bit to change whether or not I am notified of new messages in a chat. (I'll note that the recent Skype clients also have a GUI way to do this typically through a menu choice of "Chat Notification Settings...".)

I also administer a number of chats and so use /setrole to give others admin privileges and also /kick and /kickban to remove people from chats.

There are also a few other commands that aren't listed (try /info), but this list has most of the ones I'm aware of.

Anyway, try out these commands... they may help you work more quickly with Skype chats!

P.S. Users of Skype 5.0 have an additional command, "/golive", that I described in an earlier post and that provides a simple way to launch conference calls.

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Yet to be seen, of course, is how long this actually continues to work this time. Will it be an actual ongoing service? Or is this just another burst of connectivity that will fade again?

We'll see, eh?

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$("body").append(
   $("<div/>")
    .css("width","210px")
    .callme({
      apiKey: "C17D167F-09C6-4E4C-A3DD-2025D48BA243",
      numberToDial: "8007773456",
      buttonTextReady: "1-800-777-FILM",
      slideOpen:true
    })
  )

There is a Callme "demo page" online at:

http://s.phono.com/releases/0.2/samples/callme/index.htm

That shows how you could create a button with or without a dialpad and in various themes:

The source code is naturally there for you to play with if you want to do so.

Cool stuff!

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If you aren't aware of Phono, back in October Voxeo released the Phono SDK, letting you easily add in voice or chat directly into your website. The way to think about is this... traditionally, websites have had a "click-to-call" button that would call you and call a call center and bridge the two calls together. For this to work, you have to typically enter your phone number into the phone.

Phono changes that by running a softphone client directly in your web browser. So instead of entering your phone number, you simply push the button and start talking to your browser. (For those interested, we posted about the architecture, which uses a mixture of XMPP/Jingle and SIP. Phono itself is a jQuery module/library/app/whatever-you-want-to-call-it that you reference in your web page.)

Phono received a good bit of attention and we posted a lot of content about it online, including sample apps, tutorials, videos and more. The source code, too, is all available online.

One thing that always bothered us, though, was that you needed a headset for Phono to really work well... and so we spent a great amount of time working on echo suppression so that people could ditch the headset and just talk to their computer. Given that on the development side we're all Mac users, we're used to apps like Skype where you don't really need a headset. We wanted Phono to be the same.

So now we've done it... PhonoSDK version 0.2 is out! We're thrilled with how it came out and are looking forward to seeing what people build with it. If you want to try it out, simply go to Phono.com, check out the documentation and get started!

Full disclosure: In case the "we" usage above wasn't enough to clue you in, Phono is a product of my employer, Voxeo. But even if that weren't the case, I'd still write about Phono simply because it's cool... and it's disruptive.

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Ah, the dilemma of the Apple fan who does not live close to an Apple store. Unless you've been under a rock, you know that today is the day that the iPad 2 arrives... you know, it's the day that cancer will be cured, world hunger will be eliminated, all the wars will just stop, and....

... well, okay, it's really just a day when another expensive gadget comes out. Er, excuse me, a "magical and wonderful" gadget, that is. :-)

I am thinking seriously of buying one because the content creation capabilities seem vastly superior with the built-in camera... and since I won the current iPad I'm using at a conference, I'm okay buying one today.

Well, it most likely won't be today.

You see, Keene, NH, where I live, is a beautiful place to live. But it's about a 1.5 hour drive to the nearest Apple store...

The local Verizon stores have already said they won't have any iPad 2's today. We'll have to see if any other local stores do, but it's doubtful. And... when I looked online Apple is already saying that orders are 5-7 days out for shipping.

So I just may need to wait a bit... perhaps I'll hit the Apple store next week in Orlando... :-)

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Through it all, though, my ability to call Google Voice numbers via SIP kept working perfectly fine, while it stopped working (or never worked) for pretty much everyone else who had tried it. (except for one other person who saved my sanity!) Several people on Twitter thought I must have some kind of "magic"... but all I knew was that it kept on working.

Until this morning.

It's dead now.

Using SJphone to call the exact same number that has worked fine for the past two days now only gets me a constant ringing.

So whatever "magic" I may have had is gone... gone, gone, gone...

Let's hope that Google will in fact bring back this capability... and maybe even go on to provide the other side of SIP interoperability that Todd Vierling wrote about (and others have agitated around for some time). The key point being to let us point our Google Voice numbers to SIP endpoints in addition to regular PSTN phone numbers.

It was great, Google, to get a taste of SIP interconnection to Google Voice... if only for a few days. Can you please bring it back? And even make it better?

We're out here waiting...

P.S. For those not understanding the "It's Dead, Jim" reference in the title, it goes back to the original Star Trek.

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Or perhaps did work for a bit and then stopped working. Even Todd Vierling, the person who first wrote about this back on Saturday, now updated his post with this text:

[Update March 8: It seems this service is no longer working, starting yesterday evening. sigh. I was hoping to see the security bugs patched up, not for the service to be pulled down again like it was in early 2009. Please, Google Voice people, throw us a bone here and let us know what's really going on for once!]

The folks over at OnSIP published a blog post saying that sip.voice.google.com was now silent and also engaged with a whole number of us on Twitter on the topic. Ward Mundy over at NerdVittles initially came out with a post detailing how to make this work with FreePBX and then updated that post to indicate that it is no longer working.

Comments to my original post indicated some folks were having problems... and I've seen more similar comments in Twitter.

Here's the thing...

IT IS STILL WORKING PERFECTLY FINE FOR ME! :-)

Using the bare bones SJphone softphone on my iMac, I can still call into my own GV number over SIP without any problem. I can also call other people's GV numbers over SIP, too. I spoke for about 15 minutes with a friend of mine calling from my SJphone over to his GV number and reaching him on his mobile.

So it continues to work for me... but not for others. If you are on Twitter, could you perhaps let me know if you are able to call into a GV number via SIP? (Or leave a comment here to this blog post.) Perhaps if we can identify a few more folks who can call into GV via SIP we might be able to understand what's going on.

Of course, what would be best would be if Google would give us a clue about this service! Is this something experimental that people weren't/aren't supposed to know about? Is it something to be released that got out a bit too early? It's pretty obviously not production-ready...

Anyway, if you can call into GV via SIP, please do let those of us digging into it know. Would love to know the softphone you are using and perhaps the ISP you are connected to. Thanks!

P.S. My internet connection comes from Time Warner cable.

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Turns out that Skype was shooting their own video interview about the Skype / Citrix partnership which they have now posted on YouTube:

Obviously no comparison in quality to my own video interview which was shot with a handheld video camera :-) Nicely done, Skype...

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By way of a tweet from Aswath Rao (crediting @truvoip) today I learned that you could simply take your Google Voice number and append "@sip.voice.google.com" to get a perfectly working SIP URI that you could use with any SIP phone. I naturally tried it out with my own GV number using the SJphone SIP phone:

The call worked great. I answered it on one of my other phones and the conversation was fine - both audio streams intact, etc.

YATE?

What's interesting to me here is that SJphone reports that the remote client is YATE, a.k.a. Yet Another Telephony Engine. Yate has been around for a while (the voip-info wiki has some history) but hasn't been as widely known as, say, Asterisk or FreeSwitch. I subsequently made several calls using the Blink softphone on my Mac and again could see in the SIP traces that YATE was receiving the call on the Google end.

Looking over at the Yate News page, I see this note with regard to the January 31, 2011, release 3.1 of Yate:

Yate client calls can use Google Voice service.

The Yate client is a soft client for both voice and IM and in looking at their tutorial on using the client with Google Voice it would appear that this is about using XMPP (Jabber) to connect from the client over to Google Voice (I'm guessing it is using Jingle, which has been supported for some time by Google Talk (which is different from Google Voice)).

So the Yate client support is really something different... but the key point here is that Google appears to have chosen Yate to use on the receiving end of SIP calls into a Google Voice number.

WHY A SIP ADDRESS MATTERS

For some of us who have had Google Voice numbers for quite some time (mine dates back to the pre-Google-acquisition GrandCentral days), it's always been a bit frustrating that the only way to call a GV number was through the good old PSTN. Particularly because the PSTN is so... well... limiting. When I've been building apps in Tropo or Voxeo's Evolution platform, I've wanted to route them to my GV number... and I have to do this via the PSTN side. No big deal on one level, but it's just inefficient. If the call is already all on the IP side, why not just keep it all IP!

As we're off building the future of communications over IP, I've wanted to include Google Voice into that mix.

Now we can!

At least... unofficially. Perhaps at some point Google will come out and formally promote this capability.

Once a GV account has a SIP address, we then have to wonder what else we will be able to do with it. Could I, for instance, use wideband audio to my GV number?

For that to work, of course, I'd need to be able to register a SIP device with my GV number, which I can't do... and is the other side of the frustration with Google Voice. (Or at least be able to give GV a SIP URI as one of the addresses to call when a call comes in.) But conceivably once that happens I would be able to receive wideband audio calls. Ditto making video calls...

The first step is getting a SIP address that is workable... we now seem to have that.

Kudos to Google to making this inbound SIP connectivity available... and I look forward to seeing what else they will do with regard to SIP.

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