[Spce-user] having trouble with provisioning jitsi on mr3.2.1

Thu Apr 10 03:44:03 EDT 2014

Gunter,

LOL @ another one? Thanks for sharing your concerns about "my" browser
hits, quite altruistic but no, I'm definitely not confused by old tutorials
myself :)
The link I posted as suggestion was simply pulled from a previous post this
month (typo included, sorry...)

I guess we could hereby remind sporadic/new users how key is get familiar
with the mailing list archive, active population and read release
updates/documents before posting?

Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Capture Engineering
Desk: +1 (202) 470-5312
Mobile: +31 6 4603-2730

On Wed, Apr 9, 2014 at 11:28 PM, chymian <chymian at gmx.net> wrote:

> andreas,
> since another one (Lorenzo Mangani) hit the old skype-like tutorial
> yesterday, I tried to comment at the article on the correct link you gave
> me, but comments where closed...
>
> servus
> günter
>
>
> Am Freitag, 4. April 2014, 03:43:04 schrieb Andreas Granig:
> > Hi Günter,
> >
> > Thanks a lot for your feedback! You're pretty spot-on with some points
> > you raised. Yes, Sipwise is not like a small company anymore, so we need
> > to generate a significant revenue year by year to continue what we're
> > doing and to grow further. To do so, we sell commercial solutions, and
> > with that revenue, we're able to hire even more people capable of
> > supporting and enhancing the really great things we're doing in the open
> > source VoIP area.
> >
> > However, Sipwise is one of the few companies actively investing money,
> > manpower and other resources into projects like Jitsi, Kamailio and Sems
> > for those guys to be able to enhance further, strengthening the overall
> > eco-system. When you're seeing those projects announcing fancy new
> > features, they are usually sponsored by somebody, which at a great
> > extend is ourselves. This thrives the overall ecosystem, and everybody
> > is profiting from that. I was just sitting together with Emil from the
> > Jitsi project today, discussing ways how to install their sexy video
> > bridging solution in the sipwise spce in a way that it's just going to
> > work plug-and-play for all of you guys. So you probably MIGHT
> > underestimate what we're actually doing in the overall open source
> > landscape. And that MIGHT be our fault not communicating publicly what
> > we're actually up to.
> >
> > We're at this very moment at the Kamailo World Conference in Berlin
> > (which we're sponsoring to make it happen), and we're getting a lot of
> > great feedback from huge companies like Orange, who are using our open
> > source software components to serve their customers, and that makes us
> > really proud, although it doesn't actually get us any revenue.
> >
> > Having said that, we love to experiment with our own technology, and it
> > doesn't necessarily mean we're going to provide 1:1 how-tos on
> > everything we do. But you know what? It's an open source project, and of
> > course we're relying on you guys to spread it by word-of-mouth, and by
> > writing your own tutorials and how-tos on how to do stuff with the
> > system. We're more than happy to help the community out if you guys are
> > having issues with certain scenarios, but then again, we're all sitting
> > in the same boat, so the more you contribute in terms of documentation
> > etc, the more it will spread, the more people will contribute, and the
> > faster it's going to evolve.
> >
> > To answer some of your questions: Yes, one of our devs is currently
> > working on docker.io containers for spce (we're already supporting
> > vmware, virtualbox, vagrant an Amazon AWS AMIs). Yes, we just released
> > the open source rtpengine, the only highly-scalable dtls/srtp-based
> > media engine capable of interconnecting webrtc clients with the legacy
> > sip world (and basically a 100% of the people demoing webrtc at Kamailio
> > Word are actually using it), and it will soon be part of the standard
> > web interface within the CE, together with a websocket-based
> > chat-client. We just sponsored carbon-copy and jabber-search extensions
> > in Jitsi, and we're working on getting the really cool Jitsi-Videobridge
> > to be able to be used via plain SIP too. And then again, if you want
> > anything faster/earlier, get in touch with us to sponsor one or another
> > feature, or if you're serious with your customers, buy a PRO version
> > with HA and all the other fancy stuff, which directly goes back to and
> > helps other open source projects you might be using already.
> >
> > So, if it took you a while to find our project, help us reduce this time
> > for other users by spreading the word, and by writing/blogging about
> > different use cases and experiments and whatever you're doing with the
> > software!
> >
> > Andreas
> >
> >
> >
> >
> > On 04/03/2014 12:53 PM, chymian wrote:
> > > thanks andreas, that helped.
> > >
> > > and yes, I noticed that the skype-like-howto is a bit aged, but for
> getting my hands on your great software, I was looking for a howto for a
> federated sip setup - since this train is accelerating fast - and this was
> the closest I found.
> > > It would be nice, if sipwise could update this, or write an federated
> sip howto - there is a big demand out there (not only since M$ took over
> skype).
> > > i.e. I will integrate FS, video conferences, WebRTC, etc. in my
> websites (project-mgmt & collaboration/social/group-sites) since I consider
> it a must for an state-of-the-art collaboration-site. and not all the
> services will be given away free, so, there is the potential to be a future
> customer of yours, needing billing & prepaid services.
> > >
> > > in that light, I loved the "tv over webrtc" blog, but again, just a
> white paper, and no howto. I know, you as company have to make money and
> want/have to sell your solutions & expertise - on the other hand, the more
> you give, the more you will get back. especially in the OSS world, it's
> easy to to make yourself an good name and state your quality (look at
> debian) and generate money out of being known widely. and the big point,
> activate even more community and harvest the synergy.
> > >
> > > since a couple of weeks, since I was looking at federated sip (kicked
> of by danial pocock's fed. sip blog for debian) I found all types of
> solutions, but sipwise didn't appear on my radar just till the end of my
> search - after 4 weeks. and that, while I was looking exactly for a
> solution like yours...
> > > jitsi was all over the place - they have sexy solutions, well
> documented and a big community, flooding the interested ones with a lot of
> howtos.
> > >
> > > as an newcomer to the "roll your own VoIP-service" world, my
> impression so far: sipwise has a really great solution, but doesn't give
> information and howtos to the community willingly, holding back a lot of
> "sexy" possible solutions (why?) and therefore not attracting a lot of
> people, not drawing a lot of attention towards themselves.
> > > in my impression, you appear as a specialized business company, who's
> "only" interest is, to sell expertise & solutions and also "have a OSS"
> version, for the very interested ones, who are willing and can afford to
> spent very much time to dig in and set up even the easiest solutions. I'm
> aware, that the SIP/telephony-business is a complicated one, but not
> everybody want to be a full blown telco... and in my opinion, sipwise has the
> possibility to be both, a full blown telco-specialist and an easy to go,
> fed. sip provider for the "masses" - if you want that to be that, and not
> leave that marketshare to jitsi, sipexecs, a.o. (which is doing the right
> thing, in integrating it's self in clearos and do have a lot of howtos at
> hand)
> > >
> > > so on my wish list would be a debian based, zentyal.org style
> SMB-server, with integrated fed. SIP, webRTC & videoconferences, (running
> on docker.io containers), a little bit like /http://suite.tiki.org/[1] is
> doing it on centOS (w/o docker). or you even could team up with univention,
> for the money-market...
> > >
> > >
> > > just my 2 cent
> > >
> > > und danke für sipwise:CE!
> > >
> > > guenter
> > >
> > >
> > >
> > > PS: the URL has an typo, and for the mail-archive:
> > >
> https://your-ip:1443/device/autoprov/static/jitsi?user=${username}&pass=${password}&uuid=${uuid}
> > >
> > >
> > >
> > >
> > > Am Mittwoch, 2. April 2014, 11:23:08 schrieb Andreas Granig:
> > >> Hi,
> > >>
> > >> The post you're referring to is pretty old and not up-to-date anymore,
> > >> the URL you should try is:
> > >>
> > >>
> https://your-ip:1443/device/autoprov/static/jitsi?user=${username}&pass=b${password}&uuid=${uuid}
> > >>
> > >> Andreas
> > >>
> > >> On 04/02/2014 10:49 AM, günter wrote:
> > >>> BUMP
> > >>>
> > >>> anyone? please!
> > >>>
> > >>> Am Dienstag, 1. April 2014, 01:12:28 schrieb chymian:
> > >>>> hey,
> > >>>>
> > >>>> fairly new to sipwise, I could need some advice here on
> provisioning.
> > >>>>
> > >>>> the setup is mr3.2.1 with the "skype-like-howto" /
> http://www.sipwise.com/news/technical/byov-skype-replacement//[1]
> > >>>>
> > >>>> A) fresh-install on a virtual machine with a cloudserver
> jiffybox.de
> > >>>> B) used the vagrant file to UP a testmachine - just as reverence.
> > >>>>
> > >>>> on both occasions, the jitsi client fails to provision itself.
> > >>>>
> > >>>> updated jitsi already from debian version 2.4.4997-1 to the latest
> nightly: 2.5.5168-1
> > >>>> on my workstation: debian jessie
> > >>>>
> > >>>>
> > >>>>
> > >>>> 1.) when using the URL:
> https://192.168.0.122/jitsi?user=${username}&pass=${password}&uuid=${uuid}(IP changed)
> > >>>>  the jitsi config get's the first http-page from the self-care
> web-interface included. :(
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>> 2.) when I use the same URL with the provisioning port 2443, I get
> an err.-msg. in /var/log/ngcp/oss.log
> > >>>>
> https://192.168.0.122:2443/jitsi?user=${username}&pass=${password}&uuid=${uuid}
> > >>>>
> > >>>> Apr  1 00:54:47 spce apache2[6535]: [info] [client 192.168.44.1]
> Connection to child 0 established (server myserver:443)
> > >>>> Apr  1 00:54:47 spce apache2[6535]: [info] Seeding PRNG with 656
> bytes of entropy
> > >>>> Apr  1 00:54:57 spce apache2[6535]: [info] Initial (No.1) HTTPS
> request received for child 0 (server myserver:443)
> > >>>> Apr  1 00:54:57 spce apache2[6535]: [error] [client 192.168.44.1]
> File does not exist: /var/www/jitsi
> > >>>> Apr  1 00:54:57 spce oss: 192.168.44.1 - - [01/Apr/2014:00:54:57
> +0200] "POST /jitsi HTTP/1.1" 404 1290 "-" "Jitsi/2.5.5168"
> > >>>> Apr  1 00:55:02 spce apache2[6535]: [info] [client 192.168.44.1]
> (70007)The timeout specified has expired: SSL input filter read failed.
> > >>>> Apr  1 00:55:02 spce apache2[6535]: [info] [client 192.168.44.1]
> Connection closed to child 0 with standard shutdown (server myserver:443)
> > >>>>
> > >>>>
> > >>>> it's the same game on both hosts.
> > >>>> no account is created an it starts with the account-wizzard.
> > >>>>
> > >>>> do I miss something?
> > >>>> how is that supposed to work?
> > >>>>
> > >>>> ANY help appreciated
> > >>>> TIA
> > >>>> günter
> > >>>>
> > >>>> --------
> > >>>> [1] http://www.sipwise.com/news/technical/byov-skype-replacement/
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> Spce-user mailing list
> > >>> Spce-user at lists.sipwise.com
> > >>> http://lists.sipwise.com/listinfo/spce-user
> > >>>
> > >>
> > >> _______________________________________________
> > >> Spce-user mailing list
> > >> Spce-user at lists.sipwise.com
> > >> http://lists.sipwise.com/listinfo/spce-user
> > >
> > >
> > > --------
> > > [1] http://suite.tiki.org
> > >
> > >
> > >
> > > _______________________________________________
> > > Spce-user mailing list
> > > Spce-user at lists.sipwise.com
> > > http://lists.sipwise.com/listinfo/spce-user
> > >
> >
> > _______________________________________________
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> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/listinfo/spce-user
>
>
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