Topic: silentra download links

[zach_flem]

Hi folks,
i've recently started working with a new company where i will be doing a fair bit with VOIP.

at present they have been using propiertry hardware and servers which leave alot to be desired.

i have used the sme server for a few years now and would very much like to use it as a backend for what looks to be a very good voip solution.

the only thing that i cant find to help me is somewhere to download SARK!

if the full sark iso isnt an option, can someone please point me in the direction of a how-to and rpms to do an install on a current release of sme.

thanks in advance folks and i hope that i havnt missed something bleedingly obvious in my searches of the forums.


cheers

Zach Fleming
Tekace Communications
Mildura, Vic, Australia.

[JonB]

All you need is here http://www.selintra.com/docs/cgi-bin/view

including links to the rpms

Jon

[zach_flem]

cheers for your speedy reply.

will install tommorow and let you all know how i go!

thanks again
Zach Fleming

[zach_flem]

i cannot install a tftp server onto the sme box.

i have tried:

Code: [Select]

yum install tftp-server
and it returns an error saying that there is a conflict with the selintra rpm.
the same happens when i try to use another tftp server.

other things i am having dramas with is the lack of control for conferencing, the lack of information on dial plans and the numbering scheme, are we able to change the numbers used for extensions without upsetting everything else?

these are big concerns for us and need to be addressed. even if its just so we know we cant do it!



if anyone can shed some more light on this situation i'd be very grateful!

Cheers


Zach Fleming[/code]

[pmacdona]

try the following link, & follow the install procedure listed http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter03

[del]

Try this: yum --enablerepo=base install tftp-server
Worked for me

Regards,
Del

[SARK devs]

other things i am having dramas with is the lack of control for conferencing, the lack of information on dial plans and the numbering scheme, are we able to change the numbers used for extensions without upsetting everything else?

Hi

Not sure about your tftp conflict - what is the error message?

Conference control  - You're quite right - the control is not great.  However, most people manage by using a co-ordinator and the Flash Operator Panel to see what's going on.  FOP provides excellent visibility of the system and who's in what conference room at any given time.

Lack of information on dial plans - Not sure we know what you mean - can you expand on this?

Changing extension numbers - extension numbers are suggested by the system when you create a new extension.  However you can pick any 4 digit number (provided it isn't already in use somewhere else).  Yes you can change extension numbers at any time.  You won't lose voicemail (it is carried over) but you may disturb any ring or hunt groups that reference the old number.  They won't fail, but your new extension number won't any longer be part of any group that the old number may have been a member of.  We intend to put the necessary inversions into the database to flag such occurences but right now you have to track them yourself.  This is fine for a 20-100 phone switch but not good enough for a big 'un and we will fix it.

Kind Regards

Selintra

[zach_flem]

the error the server is giving me when i try to install tftp is something along the lines of "cant install <tftp server rpm> because it conflicts with <selintra rpm>"
im not at work now, so i will edit this when i get there shortly. i will also try the other suggestions people have left here (thankyou for that!)



my dial plans issue is that standard dial plans that we were playing with on trixbox dont work, we've finally found out where to actually put things and its a little annoying.

is there a list of available options when we create our dial plans?
we want to be able to mask numbers etc without having to fiddle in 3 different places (eg. dial 000 and the server appends the 0 to make it 0000)



im not so worried about changing extensions once they have been created, im more interested in changing them before i start.

most of the people around here are the brightest of poppets and they all want their new system to work just like the old one, which means 3 digit extensions and usually, starting with 1xx 2xx or 3xx with 2xx being the most common. 4 digit extensions will just confuse them way too much, and all our local numbers here in regional victoria, australia all begin with 5xxxxxxx


thanks for your help with these matters, hopefully we can get all this sorted out, we got really excited when we found faktortel was allready setup on your software as well as our spa941/2. so far we are liking this VERY MUCH!!!!

Cheers again for you time


Zach Fleming

[SARK devs]

Hi Zach,

Last first.  We've done quite a few installs over old switches with three digit extension numbers.  We just did a 40 seater with two digit items.  Here's what we did....

This client had extensions ranging from 0 (operator) thru forty-odd.  We allocated extensions of 60 + the old number so extension 20 became 6020 etc.  We also created aliases for each extension to mirror the old number (so 6020 got an alias of 20).  Aliases behave just like normal extensions so the users can dial the old numbers or the new ones and get the same result.    They can also CFWD and transfer to aliases etc etc.  Hopefully, over time, they'll get used to the new numbers but it doesn't really matter if they don't.

I'm still not getting the dial plan issue.  Can you give some examples of what you want to achieve and why you have to do it in three places?  There is only one place in the system where you can modify a number and that is in the transformation mask in Trunks.  Have a look in the docs here .... (There's actually an example from one of our sites in Oz).


http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter096#Transformation_mask  


SARK's dial plans are vanilla asterisk.  You can use any and all characters and symbols that asterisk allows.   Give us an example of what you are looking to do and we'll be happy to have a crack at it for you.

[zach_flem]

i hate to be a pain in the neck, but im having another drama, the other ones have been resolved with the help you guys have given me, or have been worked around.

now, hopefully something simple.

i've just setup everything as per the quickstart guide and everything is running fine,  except for the part where i'm entering my "global settings" changing my "country identifier" to "au" for australia and my external ip to the one that i get when i "ifconfig" my sme box.

Now that sounds all good, except for the fact that the "Operation status report" returns "Errors were found in input!"

when the screen comes back the ip has gone and the page is back to default settings with no indication of where the invalid entry is!

any help on this would be awesome!


EDIT:

Ok dont stress, i just added 3 extensions to see if i could get the internal side working without the external ip address in the global settings section, then realised i wanted to use provisioning, went in a set *JUST* the tftp server to "on" and it saved ok, tried the country identifier, all good as well, so i went thru and changed all the settings one by one, saving after each and they all worked, including the ip address, and a comit after seems to have all worked fine.

do the extensions need to be entered before the globals are set?


Zach

[SARK devs]

Hi Zach,

Order of input isn't important so it shouldn't matter when you create your extensions.  When globals gets pissed about a particular entry it usually highlights the entry in red.  What was the actual external IP address you entered?  External IP is ONLY necessary if you are running in server-only mode.  Leave it blank if you are running server-gateway.

In a commercial environment we would normally recommend that you run the switch in server-gateway mode on its own dedicated ADSL circuit.  That way you don't have to worry about big data downloads and uploads disrupting your calls and causing jitter. Even with QOS running (a lot of modem routers have their own QOS nowadays), you can still get jitter in a busy data enviroment.

Kind Regards

Selintra

[zach_flem]

Thanks again for the info.

we are running in server-gateway mode, modem in in bridge mode and server is doing our pppoe login.

should i take the ip address out? or is it safe to leave it in there?

On another note.

We are now happily using our internal phones, spa-941's and some x-lite softphones as well.

a question i have for you is about auto provisioning.

how would we go about getting our remote handsets (all of which are spa-941's) to complete their provisioning automatically.

we have a remote handset setup in-house and we would like to simulate a situation where we have scanned the "mac address" barcode on the side of the box and then handed it out to the customer, the customer takes it home, plugs it in and it goes.

do we need to load some scripts onto the phones first? or is there a way of getting them to point to our sme server's external ip for provisioning?

i have attached 2 sample scripts that i found while researching the subject of auto provisioning and wonder if we can use something similar.

the first "spa942" simply tells the phone to look for a new script using the name of its mac address and then re-sync.

the second is the final scipt which contains all the features that we want programmed into the phone.

any input is appreciated.


The addredd for the files is: http://voipspeak.net/images/stories/linksys/spa-configs.zip

cheers

Zach Fleming

[del]

Hi,

When you add the extension the spa-941 is already in the list so maybe you can choose it, enter the MAC address and in the box where it says location choose remote instead of local. I am not sure if this would work as I have never used the spa-941. I would think that somewhere in the spa-941 you would have to tell it to look for your server ie. the public IP address. Not sure, but I tend to just try things and see. Maybe Selintra can answer more in depth.

EDIT:
After thinking about it more the remote phone will probably use DHCP on the LAN that it is plugged into and we are relying on your server's DHCP to configure it. Need to think about it, there must be a way as Vonage do a simular thing with a PAP2 to give you service.

Regards,
Del

[SARK devs]

Hi Guys

Re provisioning

SARK handles local provisioning thru tftp and remote provisioning thru http.  tftp provisioning is entirely automatic provided the phone or ata arrives from the manufacturer in a condition to accept DHCP Option 66 redirects (professional stuff from people like Snom, Mitel, Aastra and Cisco does, cheaper stuff may or may not).  If it does then you can indeed just pick the phone type from the drop down and wand the mac address into the correct box and off you go. (Note  right now we do have a small problem with the 941/2 in getting it to recognise Option 66 - we're not sure why so our test 941 was set up by pointing it at the tftp server).

Remote provisioning is not so easy.  DHCP 66 is meaningless since you will not have any control over the remote dhcp server which is handing the phone its IP address.  So, the phone has to at least know where to look for its provisioned identity.  In our case, this is the external address of the SARK server (if its running in server-gateway mode) or the router protecting the net where the SARK server is (don't go there if you can avoid it - one NAT firewall we can cope with - two is just a bit greedy and you will likely get all sorts of RTP problems).  

Now, if you are Vonage or similar and you have the wherewithall, the money and the balls to order several thousand phones then most manufacturers will happily load 'em with your http provisioning server address right there on the assembly line.  Result - you just wand 'em and off you go.  For the rest of us it's a bit harder; right now you will have to unpack each unit and point it at your provisioning server as a minimum.

Sorry we can't be more helpful but that's the state of the art right now.  Other than that, your remote 941's will provision just fine.

Kind Regards

Selintra

[zach_flem]

thanks once again for your reply.

i figured that this would be the case with the remote provisioning, and for the amount of phones that we will be sending out at anyone time, i can live with the manual provisioning for the moment.

on the note of local provisioning, where on the server would we put a cfg file for the spa941's in general, so that we dont have to change anything on them at all to get them to provision? im refering to the files in my previous post (ie.  a phone specific cfg to set the mac specific cfg, which sets all the relevant options)

we are fine tuning our dial plans after you confirming the fact that its plain old asterisk, the part that was confusing us was the '_' we'd never had to do that before.

thanks for your help, hopefully i wont be causing you folks too much more drama on this subject in the future.

Thankyou for your support.

Zach Fleming