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How do you set up SLA and SCA?


mustardman
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The documentation gives just enough info to tease me to the fact PBXnSIP can monitor trunks and extensions but does not give enough info to actually set it up. SLA and SCA is the main reason I am very interested in PBXnSIP but I need to test it out to see how well it works.

 

Basically I want to emulate a Key System (SLA). Additionally, I want to share extensions (SCA).

 

SLA

I want to set up SLA (Shared Line Appearance) so that all extensions can see the status of all the PSTN lines and can barge in on the lines. If in doubt what I am trying to do just think Key system. If a call comes in on line 1 all extensions have their line 1 light flash. First to answer get's it. That extension should be able to put that call on hold at which point another extension will see the line light flashing and can pick it up. "Joe, your wifes on line 1" for example.

 

SCA

I want to set up SCA (Shared Call Appearance) which I believe would be several physical extensions sharing the same virtual extension??? This is commonly used by a secretary and her boss sharing the same extension so either/or can pick up the call. Another use is several physical extensions in one department all having the same extension. Not sure what the proper feature equivalent in PBXnSIP would be.

 

NOTE:

What I am talking about is NOT queues or ring groups or call parking etc. which I am well aware of and have a pretty good idea how to set up in PBXnSIP. They have some similar functionality but are not considered as a replacement for good ole fashioned KSU functionality that my particular small business customers are looking for.

 

As far as I can tell, this requires configuring Trunk, Account, and the Physical extension a certain way. I really don't know where to start. Any help would be greatly appreciated. An example of a setup like this would be especially appreciated.

 

I am testing with PBXnSIP on WinXP, Aastra 9133i phones, Xlite, an ITSP account and an FXO gateway. Not sure if Xlite can do this but as long as I can get the Aastra phones working I will be happy.

 

The production installs will likely have the CS410 appliance for 4 or less PSTN lines.

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Well well well.

 

SIP does not support SLA or SCA. That is sad, but true. The honesties in the IETF believe that "nobody" would need such a stupid, old-fashioned functionality.

 

So we have to live with workarounds. The workaround is "dialog state", so that someone can watch the status of an existing call. The PBX uses that for monitoring trunk lines (which we call CO-lines). You can see that someone is on the call, and you can see that it is "ringing" and if your SIP phone supports also this, you may pick up the call (using the Replaces header in SIP). When the call is parked, the PBX sends an updated XML as well that indicates a special flag that the dialog is not "rendering media" (haha). Essentially they developed a protocol for replicating the calls database. You can do the same watching with extensions and also with ACD and hunt groups. Interestingly you can do that also with night mode flags (although there is no call). I think we have done what is technically possible.

 

But it is all too complicated. Actually, the whole XML serialization stuff kills the CPU of the PBX and also of the phone.

 

We are working on a new package called "buttons" (outside of the IETF to spare us the endless discussions there), which is just used to turn buttons on and off. There is even interest from other parties like Asterisk guys. The idea is very simple: If the PBX wants to turn a "button" on, it just tells the phone to do so. No big XML hassle, just a small ASCII IM message and we are all set. This already works fine with everything that we could imagine, with the last piece of puzzle being the seizing of the shared line and the association of calls later. This protocol is also very suitable for software switch-boards that suddenly become quite lightweight.

 

So my recommendation for today is to stick to the dialog state. With that you can at least see what calls are going on in the office and that is the main point.

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Well well well.

 

SIP does not support SLA or SCA. That is sad, but true. The honesties in the IETF believe that "nobody" would need such a stupid, old-fashioned functionality.

 

So we have to live with workarounds. The workaround is "dialog state", so that someone can watch the status of an existing call. The PBX uses that for monitoring trunk lines (which we call CO-lines). You can see that someone is on the call, and you can see that it is "ringing" and if your SIP phone supports also this, you may pick up the call (using the Replaces header in SIP). When the call is parked, the PBX sends an updated XML as well that indicates a special flag that the dialog is not "rendering media" (haha). Essentially they developed a protocol for replicating the calls database. You can do the same watching with extensions and also with ACD and hunt groups. Interestingly you can do that also with night mode flags (although there is no call). I think we have done what is technically possible.

 

But it is all too complicated. Actually, the whole XML serialization stuff kills the CPU of the PBX and also of the phone.

 

We are working on a new package called "buttons" (outside of the IETF to spare us the endless discussions there), which is just used to turn buttons on and off. There is even interest from other parties like Asterisk guys. The idea is very simple: If the PBX wants to turn a "button" on, it just tells the phone to do so. No big XML hassle, just a small ASCII IM message and we are all set. This already works fine with everything that we could imagine, with the last piece of puzzle being the seizing of the shared line and the association of calls later. This protocol is also very suitable for software switch-boards that suddenly become quite lightweight.

 

So my recommendation for today is to stick to the dialog state. With that you can at least see what calls are going on in the office and that is the main point.

 

Ok.........sooooooo how do I set that up?? Nothing in the documentation.

 

Also, my demo says I am out of licenses. I only have 2 extensions+ITSP account setup. I deleted one extension (trying to get PnP to work) and now it won't let me create the second extension (giving me the "out of licenses" error). I restarted the Pbxnsip service after deleting the extension but it's still complaining.

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Can ANYONE help me. I've read a few people say "PBXnSIP is great, does SLA, I have it working and it's awesome". That is the whole reason I'm here and yet I have yet to read exactly how to do it. I'm really perplexed as to why it's such a mystery that nobody has bothered to document it yet that is the main reason a lot of people are using it. Can anyone shed some light on this please!

 

Forget about SCA, I can live without that. If someone can please just throw me a bone on the SLA thing I promise to dedicate my first CS410 install to you! :(

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Aastra also has "BLF" settings that we got working (at least the lights were changing state). I would try manual configuration here. You can also program speed dial buttons, which come handy for call pickup (e.g. program it to *87123 if you are watching extension 123).

 

Cisco: IMHO they don't support that yet in SIP. Skinny seems to be more important to them than interoprability.

 

Grandstream: Not sure. But AFAIK their extension board also supports dialog state, which is what we are doing. Manual configuration.

 

Polycom is a little bit more complicated as it is practically impossible to manually configure it (well, maybe worth a try), see http://wiki.pbxnsip.com/index.php/Polycom. There you need to use the address book and mark the entries that you want to watch. If you enter that in the extensions "Watch the calls of the following extensions" the PBX will automatically provision that for you. We did not figure out how to make them pickup a ringing call, but at least you can see that another extension is busy.

 

Snom: If you can live without Plug and Play, take a look at http://wiki.pbxnsip.com/index.php/Snom, there you can see how to use it with snom phones.

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