Wednesday, February 27, 2008

Re: [asterisk-biz] $0.0000 USA SIP to PSTN VoIP

 Dear Open source community,
 
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.
 
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 www.voipinvite.com for additional information.
 
 
 

[asterisk-biz] $0.0000 USA SIP to PSTN VoIP Termination

Frank bureau at inmte.com
Sun Jun 25 18:05:55 MST 2006 

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: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz-bounces at lists.digium.com] 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: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz-bounces at lists.digium.com] 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: > On Sun, 2006-06-25 at 20:40 +1000, Craig Lawrence wrote: > > Free calls to USA and Canada... > >  > > It looks like someone else has worked out how to interop with Skype? Or > > it's just another 'sound' business model. > >  > > http://www.icall.com/index.php > >  > > Cheers > >   > >  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 http://www.0xdecafbad.com     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 http://www.trxtel.com the VoIP provider that pays you!  --  No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.4/375 - Release Date: 25/06/2006         --  No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.4/375 - Release Date: 25/06/2006   _______________________________________________ --Bandwidth and Colocation provided by Easynews.com --  asterisk-biz mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-biz

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Canada
P 1 416 828 5262
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vijay.shan ((at)) voipinvite.com
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