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agranig
searching PlanetScale…
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31.
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Hiring by Curiosity
(sipwise.com)
1 points
by
agranig
14y ago
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2 comments
32.
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agranig
14y ago
In the good old ISDN/SS7, both the network-provided caller-id as well as the user-provided caller id are transported, even in case the caller wants to perform an anonymous call. Usually the last hop before delivering the call to the called
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agranig
14y ago
Doing a simple " svn co https://svn.java.net/svn/jitsi~svn/trunk jitsi" works perfectly fine for me. Same with the source snapshots at https://download.jitsi.org/jitsi/nightly/src/ . We also had no issues getting our bugfixes into their
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by
agranig
14y ago
The real work-horses are actually Kamailio and Sems, which are quite little known but power a LOT of your ISPs' VoIP systems. Provisioning is built on top of Apache/Perl/Catalyst with a MySQL backend. The billing system is in C and Perl, an
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agranig
14y ago
The only thing Asterisk does in our system is acting as a voicemail server.
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agranig
14y ago
The real secret sauce is their dead-simple way to sign up and use the client. As you can see in the post, the setup process with the Jitsi client is still a bit awkward, because it is supposed to be a provider-agnostic multi-protocol client
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agranig
14y ago
The Jitsi guys are currently working on an Android version, let's see what comes out of that. In general, the reason for the slow adoption of SIP beyond just pure voice telephony is that the SIP/SIMPLE standard with its companions for buddy
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by
agranig
14y ago
There is a "next steps" section at the bottom of the post pointing this out. You can check the handbook at http://www.sipwise.com/doc/2.6/spce/ar01s04.html#_creating_p... to learn how to do it.
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agranig
14y ago
The post warns about NAT on the server-side (like Amazon EC2 does). Client-side NAT works just fine.
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agranig
14y ago
One of our customers serves north of 1 mio overall provisioned subscribers on two servers with this software, so it's quite scaleable vertically already. It's easier when UDP is used though, because with TLS things add up quite quickly. To
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How to run your own open source Skype replacement
(sipwise.com)
98 points
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agranig
14y ago
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27 comments
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agranig
14y ago
You are just jealous of our plan :)
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agranig
14y ago
If you go lean, revenue is everything. You start with something small, you fight to get every single customer, you grow with every project, staffing up another developer or sysops engineer paid from your own pocket to do a bigger project th
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agranig
14y ago
Yes!?
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agranig
14y ago
Blue is so facebook, and everybody hates facebook at the moment, so we are grey/yellow for now. Has some potential for pivoting though. And since customer satisfaction is everything, we published our progress on https://twitter.com/doubles
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Ask HN: Followed all advice, why still no virality?
4 points
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agranig
14y ago
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12 comments
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Skype and how SIP is catching up
(sipwise.com)
1 points
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agranig
14y ago
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0 comments
48.
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Smart Matching in Perl with the ~~ Operator
(szabgab.com)
1 points
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agranig
14y ago
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0 comments
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agranig
14y ago
Writing great code means making it easily readable and understandable by other developers. If you want to show off your quirky syntax skills, for god's sake play perl golf or attend at obfuscated code contests.
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VoIP Traffic Analysis and Troubleshooting: the past, the present and the future
(sipwise.com)
2 points
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agranig
14y ago
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0 comments
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How VoIP works
(sipwise.com)
3 points
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agranig
15y ago
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0 comments
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How Central/Eastern Europe VCs can beat those in the Valley
(thenextweb.com)
2 points
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agranig
15y ago
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0 comments
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Sipwise raises VC investment > 1M USD for open-source VoIP soft-switch
(sipwise.com)
1 points
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agranig
15y ago
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0 comments
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When defining a new IPTV service for an ISP, then what would it be?
1 points
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agranig
15y ago
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1 comments
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agranig
15y ago
What was your reason to do both a Ltd in UK and a GmbH in Austria? We've registered an Austrian GmbH couple of years ago and haven't had any issues with that so far. The break-down of voting rights and shares are directly bound to how the c
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agranig
15y ago
This is pretty cool, thanks! Setting "double_buffer yes" prevents it from flickering in Ubuntu.
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agranig
15y ago
As for the command-line VoIP app, you can use "linphonec" included in http://www.linphone.org/ .
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Cloud Telephony - a reality check
(sipwise.com)
3 points
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agranig
15y ago
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1 comments
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agranig
15y ago
The tips at http://www.webupd8.org/2011/04/things-to-tweak-fix-after-ins... solved a lot of pain points for me.
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agranig
15y ago
Vienna, Austria Sipwise - http://www.sipwise.com Sipwise develops and integrates open-source VoIP soft-switches for Europe's biggest Cable operators. We're hiring full-time, on-site Perl developers and VoIP engineers, check http://www.si
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