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Good to see so many people there - sorry I didn't say hello to everyone! There's a copy of what I wrote on: http://unicorn.drogon.net/glug/ in several formats, so you can pick your poison... Answers to the questions that were emailled: Sam asked:
echo,destortion problems, How do you detect were the echo problems are? ISDN card, BIOS settings etc.
Echo is the hardest thing to work with and the sad thing is that you will *always* get it in one form or another when you interface to the PSTN. Echo is always caused by the far-end too, but that doesn't help us as we have to "fix" it at our end.
Several solutions - there are hardware cards - analogue, isdn2/30 which have on-board hardware cancellers. I've no experience of these.
Next is the echo cancellers built into Asterisk. You'll need to run a program called fxotune, and run it weekly acording to digium (one reason I don't buy digium hardware after being told this by their technical support)
There is also Digiums licensed HPEC - High Performance Echo Canceller - I tried this and found digiums licnesing mechanism shite, and it didn't work very well, so I dumped it.
For Zap lines (analogue and isdn30, and bristuffed isdn2), there's OSLEC. This is free and open-source, and it's fantastic. I use it all the time now. No tuning required, it "just works" in all but the most extreme cases. They are working on it with mISDN, but don't hold your breath.
For mISDN, there are parameters you can tweak in the dialstring - eg. Dial(mISDN/g:Outbound/e32:s)where the e32 is the number of taps in the echo canceller. (:s makes DTMF tones work). the e value can be 32, 64, 128 or 256. You'll need to experiment.
How do you setup Asterisk to detect settings from your SIP phone (Linksys 901), such as "Do Not Desturb" and if this phone is in a team of phones, then calls will be passed to next team member?
I don't. This is a decision I took when I designed my system - basically every SIP phone has differnet ways of doing things, their own star codes, etc. Additionally, the phone needs to be turned on for these features to work! So some phones will return "busy" when in DND mode, some will just ring, but silently. I disable phone features, if possible and do everything in the PBX.
Doing things like calling a team, passing the call down the line is a hunt-group - in my world, I build a dialplan in a loop which tries to ring each phone (or groups of phones), but before it rings the phone, it checks that phones DND flag held inside the asterisk server (and other flags line divert to voicemail) - these are set by commands (star codes) sent from the phone to the PBX.
Adrian asked:
Hardware.
I buy the telco hardware (& most of the phones) from http://www.voipon.co.uk/
Running across a VPN controlled by someone else who may not open all ports.
Company politics :) If the company needs to run VoIP between offices, then it needs to make sure it's inter-office VPN will carry it.
Asterisk machines cooperating with each other?
Use IAX. This may also help with the VPN. Although there are ways to find out how to dial remote extensions built into asterisk - eg. dundi, I've found it much easier to just hard code something - give each site a prefix - eg. for 3 sites, 2,3 and 4, then each phone in each site a 3-digit extension, them you can arrange the dialplan to Dial the right extension based on the number dialled locally -
So on site 2:
exten => _2XXX,1,Noop(Local call)
exten => _2XXX,n,Dial(SIP/${EXTEN})
exten => _3XXX,1,Noop(Call to site 3)
exten => _3XXX,n,Dial(IAX2/site3/${EXTEN})
and so on. (You need to define an IAX trunk called site2 on site 2, and
the same to get to site 4)
Well, that's one way, anyway :) Have a look at this too: http://astrecipes.net/index.php?n=204 Left Henry to last as it's the longest!
What holes do I need to poke in a network firewall: going from this, what sort of network structure would you recommend and how would you protect the VoIP from poor security, loss of privacy / eavesdropping, spam. Going on from this is the whole issue of encryption.
Firewall: SIP is port udp:5060, standard asterisk RTP ports are udp:10000-20000. IAX is udp:4569.
For normal SOHO use, a 10/100 switched network is good enough , even daisy-chaining phones to PCs (where phones have 2-port switches built-in). You may have issues with heavy workstation/server traffic, but this is rare in the average office.
There is a proposed SIP encryption thingy out there, but if totally paranoid, just run a separate physical network or use VLANs at a pinch, but some hardware can snoop multiple VLANs though.
Securuty is all about good username/passwords. Don't pick 200/200 for a username + password combination if your server is going to be exposed to the outside world! Stop spam (SPIM?) by not allowing "guest" or anonymouse remote access. (Which kills off the whole idea of being universally contactable. Ho hum)
What sort of hardware, software, providors do you like using? Follow on questions:
How can we determine who are good providors / bad providors? (A random selection of vendors include: Skype, tuxphone.co.uk, sipgate.co.uk, tesco, voipstunt, voipcheap)
Personal recomendation is the best way. Skype is proprietary and to be avoided IMO. There are other reasons too - such as running "untrusted" code inside the corporate enterprise, etc. not to mention it's firewall-busting techniques. However there are millions (?) of Skype users, so it's hard to ignore.
tuxphone is ukfsn - I didn't know JC was doing VoIP until I checked that just now. Looks like he's a Gradwell reseler at a guess, but it's hard to be sure. sipgate have been going a very long time, and even I have an account with them (no money in it though!) They use the same wholesalers as me - is that good? Who knows. Tesco - well I don't personally shop in Tesco, and I think their VoIP is expensive, but ...
Voipstunt and Voipcheap are members of a group of resellers of a company called Betamax (yes, same name as the old video format). Betamax work in a weird way - they transport the calls outside the country via VoIP, then being them back in via the PSTN. This avoids various interconnect fees, so enabled them to offer cheaper calls. Free in some cases. However the various resellers seem to change their deals on a monthly basis, so it's hard to keep up with them. Great if you're a penny pinching home user, not so great in a company IMO. I think they also use compression in their VoIP transports, so audio quality might suffer. I know at the wholesale level they offer 3 quality bands to make calls through... See the full list here:
http://backsla.sh/betamax There are many other VoIP providers - check the list on the ITSPA website. http://www.itspa.org.uk/ (Hmm. I paid my membership last month, but I'm not up there yet!)
- Hardware
- What sort of phones?
- Routers, firewalls, Analog Adapters?
- Software:
- for softphones, any issues
A lot of personal preferance here - and budget! I like Grandstream phones - cheap & cheerfull, very feature-full for the price, but maybe a little bit lightweight, and there have been software issues in the past... Linksys seem OK, but a bit more expensive - depends on whether you like Cisco or not :)
Routers, etc. You need one that can do some sort of outbound quality of service or traffic shaping. Drayteks are OK, but watch out for the 'v' models as they sort of take-over incoming SIP traffic. Their built-in ATAs are OK though.
Softphones - XLite and Zoiper are ones I've used. Not open-source though, but cross platform (Mac,Win & Linux) which is nice. Ekiga I've had a hard time working with - far too bloated IMO.
The biggest issue with soft phones is going to be the quality of your PCs audio hardware (dolby 5.1 out, but rubbish mocrophone in!), headset (you can pay more for a headset than a hard-phone!), etc.
- If I buy a VoIP number from one providor, then can I keep the number and switch providors?
Probably not. (and it's probably more technically correct to say that numbers are rented, not bought)
Number portability is the biggest headache right now. Most ITSPs can import most BT numbers, few can import Virgin numbers (Teleworst/NTL, etc.) even less can import other numbers from eg. Energis, etc. There are moves afoot to make it all happen, but it's going to take time.
Porting numbers out is going to be an even bigger headache. So we're far from being able to have a "number for life" ...
Using a PBX
Are there advantages to having your own pbx or using a centrex type service (eg both www.gradwell.co.uk and www.tuxphone.co.uk offer a £8.50 number + "unlimited" UK minutes to landlines).
One of the things I've found is that if you were to factor in the cost of your own time, then you'd stick to a standard BT line :)
If you need to talk to local PSTN hardware then you need a PBX. If you just want to make a few phone calls, then a hosted service will suffice. One issue might be the number of "internal" calls you need to make. If all behind the same broadband connection, then you might struggle with more then 4 calls - it may be that the VoIP data goes out the line, to the centrex pbx and back in again... There are ways round it, but you're fighting NAT and SIP again...
And there's no such thing as unlimited :) Read the small print.
- for PBX there is Asterisk, but there are also various flavours of this. (Freepbx, AsteriskNow, Trixbox). Any preferred implementations (or any implementations best avoided)
Trixbox has freepbx under the lid, as does pbx in a flash.Again, it's a personal thing. Any system like this will "just work" and give you something to work with fairly quickly. If that works for you, then why do anything else?
I started from scratch... Would I do that again? Not quite sure!!!
- Some websites stated that Asterisk should not be installed as(by) root. Does this matter in Debian? (For others reading this, Debian installed very easily and I had a working system running on my lan in about 30 minutes: I have not tried going outside the lan as yet)
I don't bother to run mine as non-root, but then I don't share the box with anything else - it's a dedicated appliance as far as I'm concerned...
- There seem to be two protocols involved: SIP and IAX2. What do you
recommend
and why?
- SIP has more hardware
- IAX is more modern, uses less bandwidth and is easier to set up thru
firewalls
I use what's best for the application. They both have their good points and bad. SIP is what most phones run, so use SIP for phones. SIP is used by most ITSPs, so use SIP for them too. IAX was developed to connect Asterisk boxes together, and it's quite good for that, but some people have reported that it doesn't scale very well. I use IAX to connect asterisk boxes together, and IAX to peer client PBXs back to my head end, and a mixture of IAX and SIP to connect to the PSTN (via wholesalers). I've migrating to SIP though because of bandwidth issues - I can get a client site to talk directly to the wholesalers PSTN gateway using SIP, but I can't do that with IAX unless I lose direct billing control, and that's not an option!
I have not asked the question as to whether the PBX should link directly to PTSN as frankly the hardware costs of that alone seems to be around £200+ mark; and hence seems more corporate; unless there are general issues involved.
£55.However do you have/need a PC running 24/7? In todays economic climate that might be a bigger factor than having the luxury of a PBX in the home!
So there you go. Wishing you all a happy and prosperous new year! Gordon
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