<div> Dear Open source community,</div>
<div> </div>
<div>VoIPInvite would like to offer free minutes and test bed SIP-PSTN-SIP service setups for new domestic and international asterisk PBX installations. We would like to extend all the help we can to our potential global customers. There is no commitment to buy and clients can choose any provider they like after the test. We are confident that with the premium quality and stability we provide you would stay on with us. Please keep in mind we do not provide configuration support for Asterisk, this would be the responsibility of the clients' tech staff.</div>
<div> </div>
<div><span style="FONT-SIZE: 11pt; COLOR: black; FONT-FAMILY: Verdana; mso-fareast-font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New Roman'; mso-ansi-language: EN-US; mso-fareast-language: EN-US; mso-bidi-language: AR-SA">Founded in 2005, VoIPInvite is a worldwide leader in providing integrated managed VoIP services to SMB, Enterprise and Carrier customers. It has deployed a full-featured global VoIP network utilizing switches located in Chicago, Dallas, Toronto and Europe and is trusted by more than 100 telecommunications carriers, and ITSPs worldwide. VoIPInvite terminates and originates close to a billion minutes annually. The VoIPInvite network operations center provides the reliability, security and quality of service required by the world's most discriminating customers. VoIPInvite offers SIP Trunking, SIP origination and termination services. VoIPInvite is headquartered in Ontario, Canada. Please visit <a href="http://www.voipinvite.com/"><font color="#800080">www.voipinvite.com</font></a> for additional information.</span></div>
<div> </div>
<div> </div>
<div> </div>
<div>
<h1>[asterisk-biz] $0.0000 USA SIP to PSTN VoIP Termination</h1><b>Frank</b> <a title="[asterisk-biz] $0.0000 USA SIP to PSTN VoIP Termination" href="mailto:asterisk-biz%40lists.digium.com?Subject=%5Basterisk-biz%5D%20%240.0000%20USA%20SIP%20to%20PSTN%20VoIP%20Termination&In-Reply-To=002101c698ab%24a0e6fd00%249d01a8c0%40FOCUSRCWLLJGE" target="_blank">bureau at inmte.com </a><br>
<i>Sun Jun 25 18:05:55 MST 2006</i>
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<pre>Actually.....
Just tell people to call any of the numbers below, and when prompted, enter
your iCall extension. If you're logged in, your iCall phone will ring and
let you know you have an incoming call! If you're unavailable, it'll forward
to your free voicemai
Thats more like a pbx then anythign else..
The inbound service is not direct.
-----Original Message-----
From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target="_blank">asterisk-biz-bounces at lists.digium.com</a>
[mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target="_blank">asterisk-biz-bounces at lists.digium.com</a>] On Behalf Of Craig Lawrence
Sent: Sunday, June 25, 2006 7:04 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [!! SPAM] RE: [asterisk-biz] $0.0000 USA SIP to PSTN VoIP
Termination
Whilst it's "free" they would probably also police the total minutes per
account to ensure it looks like residential use. So even if someone can
get their Asterisk server to register to multiple iCall accounts the
number of free minutes will hardly compensate them for the time in
setting it up.
I might mention the service to a call centre customer who uses
softphones to call USA / Canada. Their 100 agents could trial it :-)
Cheers
-----Original Message-----
From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target="_blank">asterisk-biz-bounces at lists.digium.com</a>
[mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target="_blank">asterisk-biz-bounces at lists.digium.com</a>] On Behalf Of trixter aka
Bret McDanel
Sent: Monday, 26 June 2006 3:27 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] $0.0000 USA SIP to PSTN VoIP Termination
On Sun, 2006-06-25 at 08:01 -0700, trixter aka Bret McDanel wrote:
><i> On Sun, 2006-06-25 at 20:40 +1000, Craig Lawrence wrote:
</i>><i> > Free calls to USA and Canada...
</i>><i> >
</i>><i> > It looks like someone else has worked out how to interop with Skype?
</i>Or
><i> > it's just another 'sound' business model.
</i>><i> >
</i>><i> > <a href="http://www.icall.com/index.php" target="_blank">http://www.icall.com/index.php</a>
</i>><i> >
</i>><i> > Cheers
</i>><i> >
</i>><i> >
</i>I got bored and didnt feel like playing with this anymore, here is what
I learned trying to get asterisk to work. Also note their softphone is
WEB ENABLED which to me, given their service model and all means they
are likely going to display banner ads. They arent in "beta" becuase
they want people to use it. This is likely why they want demographic
information to participate in the "beta", so they can do more targeted
ads.
I am guessing that is their gimick to get revenue for it. With
wholesale contracts you can likely get the per minute cost of phones
below the per minute revenue of ads, especially if you start doing 5-6
ads at the same time and rotate every 30 seconds.
The ad revenue model has been proven to not work if that is all you
have, unless these guys are doing something different, they will likely
be gone by the end of the year.
The short:
they use opensource, they appear to filter useragents and possibly other
things, they might be usable with asterisk but arent doing everything
the same way so it may be more cumbersome.
The long:
If you use their program to sign up it goes to the webserver. The web
server, their sip proxy (open ser 1.0.1-tls on x86 linux aparently
installed from generic packages) and the media proxy are the same IP. I
dont know if they are playing NAT games or not, the hostname of the box
is proxy01 however.
Their softphone has a useragent of 'iCall' and appears to be built off
sipX (LGPL).
(partial SDP from client)
a=rtpmap:0 pcmu/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:96 telephone-event/8000/1
There was reference on their forums about g.729 but that does not appear
to be in the client current as of today.
The useragent on their media gateway is 'iCall Softswitch' which means
that it could be anything. There are a few opensource softswitches out
there, and these guys appear to use opensource, so ...
They do not support silence supression either. That may be for nat
control though.
Thet appear to use port 9000 in their client for sip messages (their
server uses standard 5060 though). RTP started at 9001 (sequentially
incremented). I dont know if it randomly got 9000, but highly doubt it.
I think they opted for this in the hopes that it wouldnt collide with
other soft phones and the like.
I never received any RTP from them at all. Nor did my call go through
(ie what I called didnt ring). So I dont know if they are having some
outage or what. I did not see any STUN and the windows box is NATted so
it could be they were sending to the RFC1918 addr instead.
They are running what appears to be debian with slightly older software
(sarge ?) and what appaears to be default setups. This does not bode
well for security, given that some of what they are running has known
vulnerabilities :( I wont say what becuase no one but them needs to
know, but I will contact em about this. They are running a 2.4 kernel
too, which makes me think that its sarge as well (etch appears to always
default to a 2.6 kernel, at least in my experience). They have about 21
hours uptime, which seems odd given that they should be more stable at
this point. Unless they are playing with the kernel, who knows.
Confirmed they are running sarge, which isnt bad per se, but they do
need to upgrade some of what is on their box.
They are aparently filtering SIP traffic so if you dont appear to be
their client they wont let a call go through (they return a 401). I
dont know if this is useragent or UA+port or ...
--
Trixter <a href="http://www.0xdecafbad.com/" target="_blank">http://www.0xdecafbad.com</a> Bret McDanel
Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
<a href="http://www.trxtel.com/" target="_blank">http://www.trxtel.com</a> the VoIP provider that pays you!
--
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25/06/2006
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