[ Date Index ] [ Thread Index ] [ <= Previous by date / thread ] [ Next by date / thread => ]
Gordon Henderson wrote:
>
> 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
I would like to copy / paste the above q & a to the lug wiki, meetings
page as part of a write up for the event. Is this ok with you. I have
already put the 2 links you have sent on the page at relating to the
meeting.
thanks
Paul
--
Paul Sutton
www.zleap.net
Support Open file formats ISO 26300 odt
Next Linux User Group meet : Jan 3rd : 2pm, Shoreline Cafe Paignton
--
The Mailing List for the Devon & Cornwall LUG
http://mailman.dclug.org.uk/listinfo/list
FAQ: http://www.dcglug.org.uk/linux_adm/list-faq.html