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Re: [LUG] Elastix FREE VOIP

 

On Tue, 15 Sep 2015, Matt Stevenson wrote:

Thanks Gordon very helpful and accurate, would home users be ok on say a
1mb/s uplink ?

Budget 100Kbps *each*way* per call. For a home user that'll be fine - but some budget routers/modems struggle with the packet rate as they're optimised for large packets, not the 160 byte ones you need for VoIP.

I've run 8 concurrent calls on my own home ADSL line - upload speed 1Mb/sec. and had some customers that approached that, but for them we always had a dedicated ADSL line for VoIP only.

If you mix that with normal browsing traffic without any sort of traffic shaping then you'll end up with dalek talk. Or worse. Uploading is a killer - you have 1500 byte packets trying to be mixed with 160 byte packets... I've heard "Oh the phones stopped as soon as I sent that email" more than once )-:

I noticed you can hard code the SIP into Mitel handsets (second hand).

Not sure what you mean by that - there are 100's of handset makers that all run SIP. Many DECT systems too - great for home use as some can have the base station talking to their home analog phone line and also have a VoIP connection and you can pick which on the handsets for outgoing calls. (Gigaset)

Really appreciate the heads up on processing and transcoding that will
definitely come into it.

Have a feeling we are going to want some call metrics data reporting /
pretty graphs so that may define having a PBX of sorts rather than direct
SIP although cloud hosted could be an option.

Asterisks standard call log is nothing special, but good enough - you get call start/answer/end times... (So you can tell how long the phones rung for before being answered on incoming calls)

I did find these :
https://www.twilio.com/docs/sip-trunking/sample-configuration#elastix
https://www.voipfone.co.uk/

Also some have apps for mobiles e.g
https://www.8x8.com/

There are many. One (ex) client of mine moved to 8x8 for their hosted services but I've no idea how well it's working for them. Voipfone seem OK as far as I know. You might also want to look at http://soho66.co.uk/

Gordon


Regards

Matt



On Tuesday, September 15, 2015, Gordon Henderson <gordon+lug@xxxxxxxxxx>
wrote:
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 ânot so freeâ 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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