As I've mentioned before, it's hard to make phones sexy, especially SIP phones. The local vultures and I all had a good chuckle at that when I turned in that article, but eventually we stopped and thought: wait, how much fun can we have reviewing SIP phones? It's tough to match setting an IOsafe NAS on fire, but we have this …
This is Josh Foll
--and of the Register. T
--oday we're taking th
--e 3CX phone system for a te--
Can you guys fix that damn thing? Is there a particular reason you're trying to reinvent YouTube/Vimeo/Twitch/etc?
Re: Video fail
Right click, Save Link As...?
Android has built-in SIP now
Many users of Android phones are surprised to find out that the Android OS has has a built-in SIP phone client for some time now. It's hidden in the phone app settings under "Internet calling." It's not as full-featured as Linphone/SIPdroid/3CX, but if all you need is a basic phone, it's a lot leaner and gets the job done.
Re: Android has built-in SIP now
I have CSIPSimple on my Android. Last time I tried the built in one, the voice quality was useless. I've had an update since then, maybe it is better now, but I haven't tried it.
[offtopic] kudos for the windows taskbar config
That is also my favorite config: vertical, W2K style.
(when I'm forced to use windows, that is)
Re: [offtopic] kudos for the windows taskbar config
That's how everyone here sets up their task bars. Have since forever. The part where this arrangement is pretty crap in Win 8 is one of the reasons I'm less than keen on it.
a three-site PBX that could do unified communications would be painfully expensive.
Ah. As I'm about to embark on installing an Asterisk server in our data centre I'm a bit worried now. I've never used Asterisk, but I have some help from people who know what they're doing so I can spend my time getting comfortable with setup and config - after backup, of course. One of the things I looked at was indeed video transport - any idea how much 3CX needs in the way of bandwidth?
I'm looking at something similar to reduce our phone costs at work, but to be honest I'm having difficulty working out what parts of the stack go where, and do what.
Asterisk? SIP Trunking? DID porting?
And everyone who does a guide to VOIP porting seems to only talk about Google Voice. Which is a bit useless in the UK (and possible everywhere, as it's rumoured it's being dropped, from what I could find out).
Bit of a PITA - sorry to hijack, but anyone got any good resources for VOIP/SIP for dummies?
(running a Syno box with Asterisk)
I'd be interested in 3CX too, but baby steps, unless it's not any more difficult to implement (we have a fibre line and an ADSL line we can QOS up for it so bandwidth should be doable)
Try the FreePBX distro or PBX In A Flash. I run FPBX in a VMware VM and that's fine for quite a lot of calls/phones/users.
Do IAX to your provider. Quality is just as good and you wont end up with your phones trying to do direct audio, until you understand what the hell is going on.
When you start setting up home users, enable "Symmetric RTP" on your handsets if possible.
Disable router Application Layer Gateways for SIP - they are unneeded.
http://www.voip-info.org has some handy notes . Some is out of date.
Aastra 55i or 57i are great on site with login/out (via the Aastra scripts) and Cisco 303 or 504G are generally superb and cheap (I would heartily recommend these for an Asterisk newbie)
There is an OSS Endpoint thing in FPBX which works very well but is a little weird. Shmooze have a paid for effort. I use the OSS one.
FPBX -> Asterisk SIP Settings make sure that your **STATIC** IP address is set and that local networks is set to all internal network ranges:
172.16.0.0 err whatever the mask is
Set your extensions to default to NAT - it will work internally as well.
Don't mess with QoS until you really need to and properly understand it - end to end.
TEST, TEST, TEST - if you don't verify two way audio from a local and remote call after each change you will be caught out.
Cheers Jon, very handy stuff.
I've been fiddling with PBX in A Flash for a few nights, and still can't get my SIP clients (CSip on Android, Telephone on OS X) to authorise against the server (it has a seperate SIP trunk to a SIP provider), but I think once I manage to work out what stupid thing I'm doing wrong, I'll make much better progress.
Interesting about the IAX - I thought that was generally meant to be avoided for newbies, but it's worth a crack, eh?
Think I'll pick up a cheap VOIP handset (physical) next month once I'm paid; chances are it'll solve half my problems.
Thanks again (upvote delivered)
I had a bash at getting ENUM up and running in the UK.
The idea behind ENUM is that you dial a number in the classic way and through what are effectively reverse DNS entries, you get to a handset but via SIP/RTP directly - no telco. This is basically how you would expect IP telephony to work in the modern age.
For example a Yeovil number might be 01935 123456. You register 18.104.22.168.22.214.171.124.126.96.36.199.e164.arpa in DNS and point it at myphone.example.co.uk. Note how the number is reversed and 44 for the UK is put in and the leading 0 is dropped. Nominet is designated as authoritative for 4.4.e164.arpa - they are the UK's top level internetty registrar wossname.
So I do my research and find this: http://www.nominet.org.uk/whoweare/whatwedo/our-products-services/enum
I then sent an email asking how I sign up or whatever. It turns out that there is no register and there are no plans for one - this was around 3 months ago. A chap from Nominet also verbally (by blower) said the same - no reasons given.
So the UK have a designated registrar for ENUM IP telephony but you can't use it - end of. Conspiracy? YOU decide. OK so it would put a bit of a dent in the profits of one or two companies.
PS In case anyone is missing the point - ENUM is how IP Telephony can finally work properly in the way your Grandma would understand. At the moment you have to sign up with someone to trunk your calls - this bins that. RTP for a telephone call needs around 64Kbits-1 for reasonable quality - we all have that. SIP is a bit of a bugger to get working thanks to NAT but not insurmountable.
Bugger lost a bit of the above post through the joys of wine fuelled editing and copy n paste.
The DNS entry 188.8.131.52.184.108.40.206.220.127.116.11.e164.arpa is pointed at myphone.example.co.uk. Actually it's a little more complex than that, see http://www.ietf.org/rfc/rfc2915.txt.
Anyway, the suppression of ENUM in the UK is arse.
PS If you are having one way audio snags with SIP/RTP off site, investigate "symmetric RTP" on the handset, and disable ALGs in home routers ...
I'd thought that telnic.org had picked up that registrar role, but I'm not sure if they're still an active body.
The big issue to be fixed I guess, before this takes off, is understanding ownership of specific numbers and how they get used. It's easy to imagine lots of squatting going on and the mess that would result. Remember that any number in the UK can be owned by any telco and that assignment can change frequently. There has to be interworking between the PSTN and the Internet for this to work properly, else it will create utter chaos.
I don't know how your description of it works without a telco, or someone, selling you SIP trunks. What if the number you dial isn't an Internet destination, but a real phone? How are you going to route to that? How would you even know when placing the call?
It's a long way from being as simple as you describe. Not a conspiracy as far as I can tell, just very difficult to do properly.
Voicemail to e-mail
Really nice, 3CX Voice-mail will mail you a copy of the voice-mail left. However, it still (despite many requests and much grumbling on the bulletin board) support MP3, so it's a full-fat WAV file - not great over a poor (or mobile) e-mail link.
Other than that - probably the best SIP exchange for SOHO/SME. IMHO
Is it better than Lync?
It is? Good!
Saying that, tin cans and wet string would be better than Lync.
Although Skype is only a side issue on this subject, I have to say that recently I have found call quality and reliability to be excellent for the most part, even to Android mobiles. Maybe that becuase I'm using the Linux version and a headset? I dont know for sure other than a colleagues Vista PC setup really was crap and is now fine on W7.
Just a pity that I expect MSFT to kill it one way or another before long
Running Elastix here
Running Elastix here. It's fairly easy to do a basic setup, and IVRs, queues and pretty much anything else you'd like to do is fairly straightforward.
Currently running 160+ handsets on site, mostly wired desk phones. Up until recently we were using CSIP2Simple on our Android handsets, but the native support is pretty good now.
Have been having serious reliability issues with the Aastra 57i's. the 31i and 53i have been bullet proof, but the 57's like to randomly brick themselves. Am in the process of moving our remaining 57's over to Snom handsets, which have been lovely.
Re: Running Elastix here
I too can second Snom - Rolled out 80 handsets in a 4-star hotel and after spending time doing plenty of research beforehand, opted for FreePBX - currently atop Asterisk 1.8. The handsets are automatically provisioned via TFTP based on MAC address with a nifty DHCP Option (66 iirc). If they play up, I can remotely wipe/reprovision them without leaving my office!
The system is a Core i3, 8GB SATA SSD and 4GB of RAM with a Quad-Bri ISDN card inside a miniITX chassis for Sub-£500 (Wall mountable too!) coupled with a VoIP-Unlimited SIP trunk - The system hasn't failed once in nearly 3 years except when someone cut through a BT feed whilst excavating nearly!
I did a 1-day-training course for 3CX and it is astonishingly easy to get to grips with - Then again I had a huge advantage having only ever known asterisk beforehand. It's powerful but the licensing is rather pricey if you have a large number of users. The Snom series are a bit pricier than most but they are an absolute pleasure to use - Firmware updates are a breeze and can be automated in bulk.
Another brand to look at if on a budget is Yealink - I've just today deployed a few W52P's today (DECT IP Phones) and they're almost bang on compared to a Snom M2R feature-wise but half the price!
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