<div dir="ltr"><div><div>With Lync, you install Edge services which by default allows you to both remotely access the system (no VPN or anything extra required) and to federate with other organizations.<br><br></div>When you call direct between two federated Lync organizations it uses some flavour of SIP TLS over TCP. I've never heard of anyone trying to get that working with anything other than Lync but in theory it should be possible without much work. Lync to Cisco gateways via SIP trunking works fine so the protocol can't be all that customized/broken.<br>
<br></div><div>Sometime in the half of 2013 there is supposed to be complete Lync-to-Skype federation (audio only).<br><br></div><div>Also, Lync has always had an integrated XMPP gateway capability which allows Instant messaging between Lync and Google Talk, MSN etc. You would think they could easily build on that to add voice and video capability?<br>
</div><div><br></div>John<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Jan 28, 2013 at 12:21 PM, Adam Thompson <span dir="ltr"><<a href="mailto:athompso@athompso.net" target="_blank">athompso@athompso.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Both varieties of Skype connector existed at times.<br>
<br>
Lynx can talk regular SIP, although John will know a lot more about that than I do. (Maybe only SIP-T?)<br>
I believe you have to do some sort of directory federation to do something like the <a href="http://sip.edu" target="_blank">sip.edu</a> model... can Lynx do dynamic lookups, or does the directory object have to preexist?<br>
<span class="HOEnZb"><font color="#888888"><br>
-Adam<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
Dan Keizer <<a href="mailto:ve4drk@gmail.com">ve4drk@gmail.com</a>> wrote:<br>
<br>
>Time heals all wounds ;-)<br>
><br>
>A few years back, Digium did provide access to skype via the<br>
>asterisk-to-skype bridge product into their commercial offering ... MS<br>
>bought skype almost 2 years ago ... a month or so shortly thereafter,<br>
>Digium loses it's skype integration access and ceasing sales a few<br>
>months later -- now the skype support for skype-for-asterisk ends this<br>
>year after it's 2 year support agreement. I found this reference to<br>
>their notice : <a href="http://blogs.digium.com/2011/05/26/skype-for-asterisk-end-of-sale-july-26-2011/" target="_blank">http://blogs.digium.com/2011/05/26/skype-for-asterisk-end-of-sale-july-26-2011/</a><br>
><br>
>I'm not sure if the protocol in use to integrate Asterisk with Skype<br>
>was the Skype protocol or direct SIP, as skype provided both. Maybe<br>
>someone who did skype-asterisk integration would know...<br>
><br>
>From my understanding of Lync, all connectivity is based on the server<br>
>and it's SIP protocol extensions - so, seems restricted to me ...<br>
><br>
>Dan.<br>
><br>
><br>
>On Mon, Jan 28, 2013 at 8:21 AM, John Lange <<a href="mailto:john@johnlange.ca">john@johnlange.ca</a>> wrote:<br>
>> I didn't mean to imply that Microsoft invented the solution, just that they<br>
>> "solved" the problem by adopting the correct solution. However I do think<br>
>> Microsoft does deserve a bit of additional credit because to date they are<br>
>> the only ones to incorporate a direct sip-to-sip calling solution into the<br>
>> core of a commercial product.<br>
>><br>
>> In the case of Lync, it's becomes more viable because it's part of the<br>
>> standard implementation. If you setup Lync the standard way it's supposed to<br>
>> be installed, you get this extra bonus of being able to do sip-to-sip calls.<br>
>><br>
>> One thing I have not had any time to explore is how compatible it is with<br>
>> non-Lync deployments. Could I call someone on <a href="http://sip.edu" target="_blank">sip.edu</a> for example? As things<br>
>> stand, probably not but maybe with some work?<br>
>><br>
>> John<br>
>><br>
>><br>
>> On Fri, Jan 25, 2013 at 4:18 PM, Ron Dallmeier <<a href="mailto:rondallmeier@fiber.ca">rondallmeier@fiber.ca</a>><br>
>> wrote:<br>
>>><br>
>>> What you are describing wasn't created by MS (so far as I know)? But I<br>
>>> am happy to see that they adopted the correct solution.<br>
>>> It sounds identical to <a href="http://sip.edu" target="_blank">sip.edu</a>.<br>
>>> We had an project at MRnet (winter of 2004/2005). Bill had set this up<br>
>>> on a MRnet server (asterisk and SER). We could direct sip2sip call<br>
>>> people at the big universities in the US by their email address.<br>
>>> You attempt to make the call using the email address. The sip proxy<br>
>>> does a DNS-SRV record lookup to see if there is a sip gateway for that<br>
>>> domain, if so the call setup request to sent that way.<br>
>>> The sip gateway would know if the target user had a active sip<br>
>>> registration or the participating universities would map the email to<br>
>>> the respective campus phone using existing database lookups.<br>
>>> Funny thing the biggest problem with our project was getting it to<br>
>>> work with the UofM's PBX via PRI (we weren't getting the dialed<br>
>>> number).<br>
>>><br>
>>> ...Ron<br>
>>><br>
>>> On Fri, Jan 25, 2013 at 8:21 AM, John Lange <<a href="mailto:john@johnlange.ca">john@johnlange.ca</a>> wrote:<br>
>>> > That article about the ** thing is just a gimmick. Basically it's a<br>
>>> > speed-dial-by-name. If you a contact list you can already dial-by-name.<br>
>>> ><br>
>>> > But Ron's comment about email addresses reminded me of something...<br>
>>> ><br>
>>> > A few years back we talked about the challenge direct "dialing" people<br>
>>> > with<br>
>>> > the keypad who don't have a phone number. There have been a few<br>
>>> > proposals in<br>
>>> > this area specifically things like ENUM.<br>
>>> ><br>
>>> > What I've come to realize in the intervening time is that this problem<br>
>>> > is<br>
>>> > already well on it's way to solving itself and it's actually not that<br>
>>> > complicated.<br>
>>> ><br>
>>> > Step 1. Forget about the numeric keypad as the problem. With the<br>
>>> > proliferation of smartphones, just about everyone effectively dials by<br>
>>> > name.<br>
>>> > They lookup a contact and then press "call". On my smartphone, when I<br>
>>> > touch<br>
>>> > the "phone" button, it doesn't come up with a keypad, it comes up with a<br>
>>> > list of searchable contacts. I can't even remember the last time I<br>
>>> > called<br>
>>> > someone by punching in numbers using the keypad. So while I may not be<br>
>>> > typical, it does prove that it's possible to get rid of they keypad as<br>
>>> > the<br>
>>> > primary input device.<br>
>>> ><br>
>>> > With deskphones it's the same thing. If you look at Microsoft Lync, even<br>
>>> > the<br>
>>> > desk phone is integrated with your address book. I just start spelling<br>
>>> > the<br>
>>> > name with the keypad and typically within 3 key presses it's narrowed<br>
>>> > the<br>
>>> > list down to my intended contact (It automatically matches the name or<br>
>>> > the<br>
>>> > phone number as I type it in).<br>
>>> ><br>
>>> > Step 2. Adopt a standard for direct sip-to-sip calling. Microsoft (yes<br>
>>> > Microsoft!) has already solved this problem with an elegant and simple<br>
>>> > solution; DNS SRV. It works like this: take an email address and look up<br>
>>> > the<br>
>>> > sip service for my domain using DNS SRV, then call that destination<br>
>>> > directly<br>
>>> > bypassing the PSTN.<br>
>>> ><br>
>>> > So lets take this as a practical example. Lets say I have a contact in<br>
>>> > my<br>
>>> > address book "Mike Smith". I probably have his name, email and phone<br>
>>> > number.<br>
>>> > If I want to call Mike, I look up his contact and touch "call". In the<br>
>>> > background my voice application (my "phone") does DNS SRV lookup for SIP<br>
>>> > using his email address. If it returns a result, I "dial" that and talk<br>
>>> > to<br>
>>> > Mike. If it fails, I fall-back to a traditional PSTN/Cell call.<br>
>>> ><br>
>>> > John<br>
>>> _______________________________________________<br>
>>> Asterisk mailing list<br>
>>> <a href="mailto:Asterisk@muug.mb.ca">Asterisk@muug.mb.ca</a><br>
>>> <a href="http://www.muug.mb.ca/mailman/listinfo/asterisk" target="_blank">http://www.muug.mb.ca/mailman/listinfo/asterisk</a><br>
>><br>
>><br>
>><br>
>><br>
>> --<br>
>> John Lange<br>
>> <a href="http://www.johnlange.ca" target="_blank">www.johnlange.ca</a><br>
>><br>
>> _______________________________________________<br>
>> Asterisk mailing list<br>
>> <a href="mailto:Asterisk@muug.mb.ca">Asterisk@muug.mb.ca</a><br>
>> <a href="http://www.muug.mb.ca/mailman/listinfo/asterisk" target="_blank">http://www.muug.mb.ca/mailman/listinfo/asterisk</a><br>
>><br>
>_______________________________________________<br>
>Asterisk mailing list<br>
><a href="mailto:Asterisk@muug.mb.ca">Asterisk@muug.mb.ca</a><br>
><a href="http://www.muug.mb.ca/mailman/listinfo/asterisk" target="_blank">http://www.muug.mb.ca/mailman/listinfo/asterisk</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>John Lange<br><a href="http://www.johnlange.ca">www.johnlange.ca</a>
</div>