ClearFoundation

First off, WOW!

Can't thank everyone in the community, admin and developer's team enough!

Everyone works harder here than at your typical main stream network (Hardware and OS) venders!!

I really got hell bent on trying this new version when I saw the screen shot that had a VOIP tab. I've been unable to locate any "planned" implementation notes on having a functional VOIP built into the OS, but my god if its built into the gateway device I could see this going platinum faster than ever before.

I am rather new to Linux, but I dropped learning Russian to become fluent in it.

So I guess my question is two fold, when will the test sever be up? and Is a VOIP server going to be built into the OS?

Hi tillburn,

Vercom will be providing a VoIP/PBX solution that is integrated with ClearOS. More information will be made available as we approach the 5.1 final release. In the old ClarkConnect, they have both a free lite version (no support of course) and a full version. That's just the beginning. I am sure there will be other VoIP/PBX solutions coming to ClearOS over the next couple of years.

I have played around with Asterisk on a number of occasions. I can say with some authority that I am a rather (ahem) seasoned system administrator who knows a lot of tips and tricks when it comes to ClearOS. Asterisk (even with an easy-to-use web interface) is one of those applications where you either have to:

1) put in a lot of time and resources to figure it out and maintain it, or
2) outsource it to an organization to install and maintain it.

From echo cancellation to esoteric phone menus to "I didn't know that" moments... that's Asterisk. VoIP solution providers have their own world outside of server/gateway software. They speak a different language.

Things like SugarCRM, Joomla and MythTV (my personal favorite!) fall into a similar category. Yes, you can get these applications up and running, but be prepared to spend significant time with them if you don't have someone supporting you along the way. If you get stuck, it's often nice to have someone knowledgable on the other end of the phone to get you of a jam... especially for something critical like a phone system!

Woah... I digressed there.

If you have not yet ventured into asterisk I assure you that though there is an initial hurdle, once the basic concepts and syntax is learned it is not terribly difficult. I felt that asterisk was a big and complex project but I now consider myself a seasoned asterisk admin and can install a basic setup with asterisk and freepbx on a debian or centos box from scratch without issue.

syadnom wrote:
If you have not yet ventured into asterisk I assure you that though there is an initial hurdle, once the basic concepts and syntax is learned it is not terribly difficult. I felt that asterisk was a big and complex project but I now consider myself a seasoned asterisk admin and can install a basic setup with asterisk and freepbx on a debian or centos box from scratch without issue.

It's good to hear that there is someone around that know something about asterisk, I have a question about cards, I understand that you need a analog card to plug phone lines in too, but what are the interfaces that come on the card and what do they do, ie,, trunk and station interfaces, and does this analog card handle faxing or is that a different card?

What other cards will you need to do VIOP, this is where I get lost every time, I can't seem to find good basic information on this subject everybody just seems to be selling something, and where I do find the information there not enough information to pick a card from the hundreds of cards to choose from?

So if you would not mind maybe you could give use a run down on the hardware a little.

pete

There are two interface types, FXO and FXS

O = POTS lines and S = Analog Stations.

For faxing you use an FXO port for the incoming line and an FXS port to plug the fax into. You can also use ATA devices which put an FXS port on the end of a SIP extentions. You must be carefull with SIP and faxing and use the right codec(ulaw or alaw) and must have an interface card with timing circuitry.

You can also interface with T1 cards. If you want to fax over the T1 then you need to have a seperate FXO card plugged into a POTS line and run a timing cable over it typically.

I suggest Sangoma A200 cards. They are modular and expandable and can house box FXO and FXS ports on the same card.

Other hardware is just phones. You can even skip the FXO/S/T1 inteface cards and go 100% SIP if you like and even virtualize the phone system. I typically virtualize test machineas and use a grandstream ATA device to get my POTS lines into the virtualized setup. These ATA devices may be a SIP <> FXS or a FXO <> SIP or a combination of both. Grandstream makes a few of these and they are handy.

ATA examples
Grandstream HT503 - 1FXO and 1FXS www.voipsupply.com/grandstream-ht-503
HT502 = 2x FXS www.voipsupply.com/grandstream-ht-502
HT486 = 1FXS www.voipsupply.com/grandstream-gs-486

What is super nice about these devices is that they are actually a little NAT router so you can put a phone or fax on any public IP and have a fax on the FXS port and plug a phone into the LAN, then plug a PC into the phone giving a complete home office setup with a SIP phone, fax machine, and computer connection.

If you are just trying to test the VoIP waters, then just get PBX in a flash and a HT503. You will have to use software echo cancelation as these ATA devices either dont have it or it isnt terribly good but the software echo cancelation in asterisk is quite good.

more next post..

Codecs are another fun detail. Basically you get g711 with ulaw and alaw flavors where ulaw is the US and alaw is everyone else with any system which is uncompressed PCM. It is good for networks with some packet loss and/or some latency but uses 64kb/s for data and then another 30 or so in protocal overhead. You can use something like iLBC or GSM for free also which cuts the data requirements down to 13-14kb/s but still has 20-30kb/s in SIP protocal overhead.

Faxing only works over g711 only as it is non-compressed. any compression codec nukes faxes basically because a lost packet causes a 'fill' where the gap in audio you might here is replaced by replaying the previous sound. There is t38 faxing which helps but just stick with g711 for faxing.

I like to use ulaw for any and all local calls. I also like ulaw to my SIP provider when I have the bandwidth available. With remote workers I usually do iLBC because they are likely to be using a softphone on their laptops and using an aircard which likely to drop packets so I need low bandwidth and resiliency to packet loss which iLBC handles pretty well. GSM is also decent. You could hear a little bit of interfearance but this is typically acceptable.

For phones I am a die-hard Aastra fan. I like their 55i and 57i phones a lot and deploy them almost exclusively. www.voipsupply.com/aastra-55i www.voipsupply.com/aastra-57i
I should add that there are a lot of good phones out there like Polycom (great audio quality) and Snom. Granstream makes some very nicely featured phones but the build quality of their phones is reflected in the price. I like Aastra phones because they have an XML browser on them and you can do things like visual voicemail, company directory, google search, weather, etc etc.. They also can be provisioned very very easily as you can log into a phone on an established extension without having to program the phone individually.

For interface cards, go with Sangoma. They are the best. Best build, best sound. Rhinos are good also but dont sound as good. I dont like digiums cards much as I have had problems with multiple cards in one box. generics are garbage. I would do a grandstream as an FXO before going with a generic hardware card.

you may also notice my vendor of choice is voipsupply. Their sales rep Arthur has been great and their prices very good. www.voipsupply.com/

Hi,

What has happened with this ? is there going to be asteisk support on ClearOS anytime soon ?

Also I have been trying to find the screenshots mentioned here where it shows a VoIP tab but the ones on Software->ClearOS don't have anything related to that.

Thanks

I would be interested in some level of support.

considering the base of clearos is centos and there is solid support for asterisk and freepbx as well as rpms for all the major interface cards it would not be difficult to install a PBX on top of clearos. Dont re-invent the wheel by working a seperate interface, just put a freepbx tab on the web interface for access and some firewall defaults to allow the SIP traffic.

@Juan

It's coming. I just connected with Vercom yesterday and are making progress with the ClearOS 5.1 version.

@Synadom

There has never been quite enough developer interest to get FreePBX going. There are a couple of howtos floating around, but a nice and easy install has never been done. Whoever takes on the responsibility is going to:

1) Have to use it regularly
2) Be prepared to maintain the software and provide community guidance

Since FreePBX and Asterisk are complex (compared to something like UPnP) it's a big commitment!

Thanks for the answer Peter, just a couple more questions about this:

1. Do you know an estimate of when could it be ready?
2. It is going to be a freely downloadable add-on ?