Topic: [Announce] SAIL-2.1.11-142 Beta

[SARK devs]

Hi Everyone,

SAIL-2.1.11-142 is now available for download from our ftp site at;

http://selintra.com/docs/cgi-bin/view/Main/DownLoadPages

This is a biggish release with a lot of new functionality so you will almost certainly find bugs in it.

Highlights include;

Full, Multi-Level IVR
Custom Applications
Fully automatic TDM/X100 board detection/configuration
chan-spy implementation
Message management panels
Updates to the templates for phones and carriers
Simplified Sipura ATA3000 support
General panel cleanup and bugfix

This release has been tested with the rc1 releases of asterisk (asterisk-SME7rc1), we have not tested it with any earlier releases so if you are running SME7 pre rc1 you may want to wait.  There's no particular reason why it shouldn't run but just be aware that we haven't done any testing here.

Next out will include ISDN BRI support, probably EICON DIVA Server as a minimum.

Kind Regards

Selintra

[poe]

Just wantet to say youre doing a great job on with SAIL.

I have only one small problem that has to do with asterisk itself.

Somewhere along the way (updates/reinstalls) asterisk stopped starting at boot time.

Could you tell me the best way to make it start at boot time, while maintaining compability with the asterisk rpm you use?

If you should have some spare time one day - a description of best practice when upgrading SAIL would be nice. My server complains about the existing database when doing a rpm -Uvh.

Best regards, and thanks again.

Preben

[SARK devs]

Somewhere along the way (updates/reinstalls) asterisk stopped starting at boot time.

Hmmm sounds like you've lost a link.  We've seen this happen a couple of times but haven't manage to track down the reason why.

Do

Code: [Select]


ls  /etc/rc.d/rc7.d

and look for S93asterisk.

I suspect its not there.

If it isn't, do

Code: [Select]


cd /etc/rc.d/rc7.d
ln -s /etc/rc.d/init.d/e-smith-service S93asterisk


That will cure your start-up problem.

My server complains about the existing database when doing a rpm -Uvh.

Don't worry about this, it's benign.  We just weren't smart enough to figure out whether the database had been created or not in the rpm.  I think they fixed this in the later releases but I'm not 100% sure.  Either way, it's not a problem.


Regards

Selintra

[xboxer21]

How do we upload a greeting?

Thanks

[SARK devs]

How do we upload a greeting?

Go to any phone and dial *60*nnnn (where nnnn is the number of the greeting you want to record).  Follow the voice instructions.  You'll have to input a password - it's the "4-digit Password for KEY Ops:" in Globals panel, default is 1111.

Once saved it will appear in the greetings panel and you can optionally give it a description.

If you have a sound recording you've made externally then you can save it directly in

/var/lib/asterisk/sounds

as a .gsm file with the name

/var/lib/asterisk/sounds/usergreetingnnnn.gsm

where nnnn is the greeting number.


For a complete list of keypad operations see

http://selintra.com/docs/cgi-bin/view/Main/SysKey

Kind Regards

Selintra

[xboxer21]

Thanks Selintra

[dswillia]

Excellent Job guys, this last beta actually made me dump AMP.  I do have a couple questions though.

It seems as though calls are automagily answered and put into a queue and then ring to the operator extension.  I have 3 GXP 2000's I would like to all ring at the same time, no operator.  Is there a way to do this without having to create a queue and logging in all the users from the phones?

Also is there a way to pass caller id to the incomming calls, as of right now my gxp's are just showing the name of the pbx (account name)  This was working fine in AMP

Regards,

[SARK devs]

Hello ds - glad you like what we're doing.

I have 3 GXP 2000's I would like to all ring at the same time,

Currently, this isn't available. Our view (perhaps wrongly) is that group-pickup and CFBS do this better than group ring, which can be distracting in an office environment.  Here's how it works;

Make sure all three phones are on the same pickup group (set in extensions panel). Put the phones into a CFBS loop (i.e. 1->2->3->1) using *22*. You shouldn't do this with most asterisk implementations because they've got no loop/spiral detection, however it's perfectly safe with SAIL.  Point your inbound trunk at any one of the 3 phones.  An inbound call will now always ring at least one phone (uness all three are busy, in which case, it will drop to voicemail) .  The call can be picked up from any phone using group pickup (*8#).  If this is not practical for you (perhaps the phones are not within earshot of one another), then we will put ring groups in for you.  It's not difficult to do, we just haven't done it yet.

Also is there a way to pass caller id to the incomming calls

Can we ask a few questions to understand this better?  

Can you explain what you mean by "name of the pbx (account name)"?

What exactly are you seeing on the GXP2K screen?

Where are the calls coming from?  Are they VOIP or analogue, and if analogue, how are you interfacing to the analogue line (TDM board, Spa3K, whatever).

Thanks in advance.

Selintra

[Tib]

ok

I think I need a bit more help here ... I recorded a greeting but have no idea how to get the system to play my greeting and not the standard one.

I have my trunk lines setup ... Sipura 3000 and Astratel

I have Extentions all setup ... spa1 on 5000 and two X-Lite 5001 and 5002

I have 2 routes setup .... one to go through the sipura3000 and one to go through Astratel.

I have a heap of speed dial numbers

I have setup an Agent ... not being used atm

Also have one queue called general ... not being used atm

I have recorded a Greeting that I called ...Standard greeting

To get this greeting working do I setup IVR menus?

I have no idea what way to go atm.

Maybe a flow chart of what to setup for where and what ties in with what could be good for ppl like me

Regards,

Tib

[dswillia]

Thanks for the response, my problem is that I run this in my home, and the phones are not withing earshot, ring groups would be nice, however queues seem to work fine so don't spend any "development" time on my account.

Can you explain what you mean by "name of the pbx (account name)"?

It just says Incomming call, rings 2 lines and says "asterisk asterisk" on the CID.

I am accessing the TDM lines via a X100P "clone" card, as well as a Voip Circuit from MixNetworks.  Neither display CiD info.

Also, generaly when running amp/asterisk on the server I had to open ports 5060 and 10000 to (Whatever) UDP for my system to get incomming voip calls.  Does SAIL do this for me, or do I still need to open them?

Regards,
dswillia

[SARK devs]

Hi Tib,

Sounds like you've been busy.  You don't have much more to do to activate your greeting.  Go to the IVR panel and create a new menu.  Name it and in the greeting drop down you will see all of your user-recorded greetings.  Choose the one you want to use for this menu.  We assume your greeting asks the caller to make a choice of some kind based around hitting keys on the phone.  In the menu you can specify the required  actions against the various keys.  The drop downs contain all of the extensions on your system together with all of the "action" entities you've defined.  An "action" entity can be a menu, a custom app or a queue.   Against the keys you've referenced in your greeting choose an action or an extension to call.  In the "Timeout" box you also choose an action or extension  (which will occur if the caller does nothing).  

To publish your greeting choose the trunk for which you want it to be active and press the change (CHG) button.  On the change panel, turn IVR on (it's a check box) and choose your greeting number from the "open Greeting number" drop down.

That's it you're done.  Call the trunk and test your greeting.

Regards

Selintra

[xboxer21]

I'm having trouble configuring SAIL to work with Teliax and broadvoice.
If anybody here has sucessfully configured sail please post the how to.

Thanks

[SARK devs]

Also, generaly when running amp/asterisk on the server I had to open ports 5060 and 10000 to (Whatever) UDP for my system to get incomming voip calls. Does SAIL do this for me, or do I still need to open them?

See

http://selintra.com/docs/cgi-bin/view/Main/AstUdpPorts#SME_Server_Version_7_0_and_UDP_P

for a discussion of UDP port issues.


Regards

Selintra

[Tib]

Thank you Selintra,

That worked nice ... one thing though once the extention rings out it goes back to the standard asterisk message ... that would be ok if it actually said something about leaving a message but it doesn't.

Is there a way to over write that message or put a different message in place of it.

Also how do I get the system to ring a few more times then what it does ... it's a bit too short if your not near the phone.

Regards,

Tib

[SARK devs]

how do I get the system to ring a few more times then what it does

Hi Tib,

It's set in Globals (Ringdelay) - see

http://selintra.com/docs/cgi-bin/view/Main/SysGlobals2

Is there a way to over write that (voicemail) message or put a different message in place of it.

Yes, it's actually pretty sophisticated.  You can read a full discussion here

http://www.asteriskguru.com/tutorials/asterisk_voicemail.html

To record your busy and unavailable greetings go into voicemail (*50*) then press 0 (mailbox options) followed by 1 to record your unavailable greeting or 2 to record your busy greeting.  


Kind Regards

Selintra