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Re: [LUG] Elastix FREE VOIP

 

On Tue, 15 Sep 2015, Matt Stevenson wrote:

That model of plugins / support bolt ons sure adds up Simon as you say. $70
per hour for just chat support.
Seeing this with model emerging more on ʽnot so freeʼ hidden costs
throughout some sites.
Even freepbx wants $150 for 1 hour.
We found similar with WHMCS and support + modules.

So we may need to shop around as we are looking for pbx that will support
home workers a small team under 10 people.

Be good to hear experiences of this.

I've thought about open sourcing my own (asterisk based) PBX solution - however the down-side is that people would want hand-holding during the install then even more help to keep it going - so the only way to sustain this would be to charge for support... Them it's no different from the above.

So you get what you pay for...

There are many hosted PBX solutions out there. You just need to find one that's affordable for you - and allows you to choose your own outgoing and incoming SIP providers. (if you want that)

Or just rent an amazon VPS and put freePBX on it...

But first; google sipvicious

All PBXs on the internet get hit by thieves trying to break into them with the intention to steal your precious VoIP call minutes (which translates to money) I've seen sipvicious attacks last for days and days - and while you can firewall them off, older versions of sv do not stop - they still send packets into you and if you have to pay for that, then you have to pay for that. I've seen ADSL lines maxed out for days and days and the end-users having to pay for line top-ups (on quality data limited services)

A PBX to support 10 people using asterisk can be run on a P100. I benchmarked mine at 200 extensions and 100 concurrent calls and that wasn't an issue on a VIA 1GHz processor. (It degraded non-linearly after about 120 concurrent calls)

Transcoding needs cpu cycles (so don't do it, but do be aware than Asterisk likes to transcode to GSM when recording calls & voicemail unless told otherwise) but pushing packets round doesn't - but does need a good Ethernet system (so don't use a Pi for anything other than trivial use) One VoIP call is 50 packets per second in and 50 pps out. A PBX needs to handle 2 "legs" (pbx to phone and pbx to SIP provider) so effectively 2 calls per single call - that's an immediate load of 200 packets per second (160 byte packets + IP overhead) There are ways to hairpin SIP provider directly to handset - and SIP was designed to do just that, however NAT gets very much in the way of that unless you want to spend a lot of time & money fixing it.

Gordon
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