WEBVTT 00:00:00.000 --> 00:00:03.423 align:middle line:90% 00:00:03.423 --> 00:00:06.846 align:middle line:90% [AUDIO LOGO] 00:00:06.846 --> 00:00:09.790 align:middle line:90% 00:00:09.790 --> 00:00:12.145 align:middle line:90% Hello, and welcome to INE. 00:00:12.145 --> 00:00:14.980 align:middle line:84% Today in this video, we will look at section 2 part 4, 00:00:14.980 --> 00:00:17.320 align:middle line:90% basically, task 2.8. 00:00:17.320 --> 00:00:19.420 align:middle line:84% Now, you understand it says use the cube 00:00:19.420 --> 00:00:24.370 align:middle line:84% to limit calls from HQ or site B to the SIP ITSP, which 00:00:24.370 --> 00:00:25.190 align:middle line:90% is the PSTN. 00:00:25.190 --> 00:00:28.690 align:middle line:84% So if HQ calls PSTN or if site B calls PSTN, 00:00:28.690 --> 00:00:32.680 align:middle line:84% we need to limit the amount of calls on the queue. 00:00:32.680 --> 00:00:35.650 align:middle line:84% Now, if you remember the cube was R1 and R2. 00:00:35.650 --> 00:00:38.740 align:middle line:84% We had configured the application redundancy 00:00:38.740 --> 00:00:40.070 align:middle line:90% in one of the earlier tasks. 00:00:40.070 --> 00:00:42.310 align:middle line:84% So we need to limit the amount of calls now. 00:00:42.310 --> 00:00:45.880 align:middle line:84% It says configure the cube so that only one simultaneous call 00:00:45.880 --> 00:00:52.180 align:middle line:84% is allowed to or from the ITSP and any additional calls should 00:00:52.180 --> 00:00:58.130 align:middle line:84% basically return a busy signal with SIP 486 response. 00:00:58.130 --> 00:00:59.590 align:middle line:90% So let's quickly go and do this. 00:00:59.590 --> 00:01:01.180 align:middle line:84% Now before we configure this, let's 00:01:01.180 --> 00:01:07.510 align:middle line:84% make an outbound call from HQ to PSTN and from site B to PSTN. 00:01:07.510 --> 00:01:09.070 align:middle line:90% Let's make that outbound call. 00:01:09.070 --> 00:01:11.830 align:middle line:90% 00:01:11.830 --> 00:01:18.870 align:middle line:84% So what I'm going to do is I will go to my HQ call manager. 00:01:18.870 --> 00:01:23.190 align:middle line:84% I'll open up site B as well here. 00:01:23.190 --> 00:01:25.530 align:middle line:90% And let's go to Jabber. 00:01:25.530 --> 00:01:31.380 align:middle line:84% So H Jabber1 has Tony Stark, which is a user on HQ site. 00:01:31.380 --> 00:01:36.090 align:middle line:84% Let's go and also access Jabber2. 00:01:36.090 --> 00:01:38.570 align:middle line:90% So Jabber2 is here. 00:01:38.570 --> 00:01:41.370 align:middle line:84% And let's also open up Jabber3, which is the PSTN. 00:01:41.370 --> 00:01:48.700 align:middle line:90% 00:01:48.700 --> 00:01:53.440 align:middle line:90% So let's close this. 00:01:53.440 --> 00:01:56.510 align:middle line:90% Say OK. 00:01:56.510 --> 00:02:03.490 align:middle line:90% And let's go back to Jabber1. 00:02:03.490 --> 00:02:06.990 align:middle line:90% 00:02:06.990 --> 00:02:12.480 align:middle line:84% And let's do an outbound call to PSTN, 95151111. 00:02:12.480 --> 00:02:15.630 align:middle line:84% Now before I make the call, let's go to R1 00:02:15.630 --> 00:02:17.950 align:middle line:90% and do a debug CCSIP messages. 00:02:17.950 --> 00:02:21.670 align:middle line:90% 00:02:21.670 --> 00:02:29.460 align:middle line:84% So debug ccsip messages, and let's make that outbound call. 00:02:29.460 --> 00:02:32.940 align:middle line:84% You should see that I should see a debug here, 00:02:32.940 --> 00:02:36.210 align:middle line:84% and the call should ring on PSTN, 00:02:36.210 --> 00:02:38.460 align:middle line:90% and let's answer this call. 00:02:38.460 --> 00:02:43.680 align:middle line:84% Let's also go to Jabber2, which is site B, 00:02:43.680 --> 00:02:50.820 align:middle line:84% and make an outbound call to PSTN again, so 95252222. 00:02:50.820 --> 00:02:55.950 align:middle line:84% So you will make one more call to the PSTN. 00:02:55.950 --> 00:03:01.440 align:middle line:84% We should see the phone ringing here as well for the PSTN. 00:03:01.440 --> 00:03:05.070 align:middle line:84% In fact, let's go to the cube and see. 00:03:05.070 --> 00:03:08.540 align:middle line:90% So I see trying here. 00:03:08.540 --> 00:03:11.840 align:middle line:90% Let's go back to PSTN. 00:03:11.840 --> 00:03:15.650 align:middle line:90% 00:03:15.650 --> 00:03:17.920 align:middle line:84% So I see Steve Rogers is also calling. 00:03:17.920 --> 00:03:22.810 align:middle line:84% So two calls were basically working. 00:03:22.810 --> 00:03:24.530 align:middle line:90% Now we want to restrict that. 00:03:24.530 --> 00:03:29.950 align:middle line:84% So for restricting that, I would go back to my cube, 00:03:29.950 --> 00:03:33.490 align:middle line:90% and let's in fact end the call. 00:03:33.490 --> 00:03:35.590 align:middle line:84% Or we can keep this call running, 00:03:35.590 --> 00:03:38.890 align:middle line:84% or you can end the call and then configure it and then retest. 00:03:38.890 --> 00:03:43.330 align:middle line:84% So let's go back to R1 and do config t. 00:03:43.330 --> 00:03:47.620 align:middle line:84% The first command that we would give to do some restriction 00:03:47.620 --> 00:03:57.020 align:middle line:84% is enable the service, which is done with call treatment on. 00:03:57.020 --> 00:04:00.580 align:middle line:84% So if you look at the options, we have on. 00:04:00.580 --> 00:04:02.840 align:middle line:90% This is by default disabled. 00:04:02.840 --> 00:04:05.560 align:middle line:84% So enable that, so call treatment on. 00:04:05.560 --> 00:04:10.120 align:middle line:84% Then we will say call treatment cause-code. 00:04:10.120 --> 00:04:12.190 align:middle line:84% And we have a list of cause codes available, 00:04:12.190 --> 00:04:14.050 align:middle line:84% because the question states that we 00:04:14.050 --> 00:04:17.529 align:middle line:84% need to send the SIP busy message, which 00:04:17.529 --> 00:04:21.430 align:middle line:84% if you look at the question, it says we need to send a SIP 486 00:04:21.430 --> 00:04:23.590 align:middle line:84% response, which is a busy signal that's 00:04:23.590 --> 00:04:26.020 align:middle line:90% going to be cause-code busy. 00:04:26.020 --> 00:04:28.450 align:middle line:84% And then we will define the threshold, 00:04:28.450 --> 00:04:34.310 align:middle line:90% so call threshold global. 00:04:34.310 --> 00:04:35.640 align:middle line:90% Let's do a question mark first. 00:04:35.640 --> 00:04:36.900 align:middle line:90% So we have some options here. 00:04:36.900 --> 00:04:38.990 align:middle line:84% So we can define it for interface, 00:04:38.990 --> 00:04:40.863 align:middle line:90% or we can define it globally. 00:04:40.863 --> 00:04:42.030 align:middle line:90% So let's define it globally. 00:04:42.030 --> 00:04:45.320 align:middle line:84% So irrespective whether the call is going outbound or coming in 00:04:45.320 --> 00:04:47.040 align:middle line:84% from PSTN it doesn't really matter. 00:04:47.040 --> 00:04:51.330 align:middle line:84% So call threshold global, total amount of calls. 00:04:51.330 --> 00:04:54.770 align:middle line:84% So we want to restrict the total amount 00:04:54.770 --> 00:05:02.000 align:middle line:84% of calls, that's, total calls, low as one and high as one. 00:05:02.000 --> 00:05:03.770 align:middle line:90% That's all we need to do. 00:05:03.770 --> 00:05:06.540 align:middle line:84% The same thing we would do on R2 as well. 00:05:06.540 --> 00:05:08.690 align:middle line:84% Make sure you do not forget to do it 00:05:08.690 --> 00:05:12.260 align:middle line:84% on R2 because the configuration is not replicated. 00:05:12.260 --> 00:05:15.090 align:middle line:84% So you would have to manually do it here as well. 00:05:15.090 --> 00:05:23.630 align:middle line:84% So call treatment on, then call treatment cause-code busy, 00:05:23.630 --> 00:05:29.570 align:middle line:84% and then call threshold, total or global, 00:05:29.570 --> 00:05:35.000 align:middle line:84% total amount of calls, low would be one and high would be one. 00:05:35.000 --> 00:05:36.650 align:middle line:90% Let's save this. 00:05:36.650 --> 00:05:38.990 align:middle line:90% Go back to R1. 00:05:38.990 --> 00:05:41.840 align:middle line:84% Let's save this, and we have a debug turned on. 00:05:41.840 --> 00:05:49.320 align:middle line:84% Let's try and make the call from HQ site again, so 95151111. 00:05:49.320 --> 00:05:51.770 align:middle line:90% This call should work. 00:05:51.770 --> 00:05:56.780 align:middle line:84% It should ring on the PSTN, and which it does. 00:05:56.780 --> 00:05:57.950 align:middle line:90% Let's answer this. 00:05:57.950 --> 00:06:02.450 align:middle line:84% Now if site B calls, site B should not be able to call. 00:06:02.450 --> 00:06:06.140 align:middle line:84% The cube should send a response back to SME, 00:06:06.140 --> 00:06:10.830 align:middle line:84% stating that it's busy 486 response. 00:06:10.830 --> 00:06:16.305 align:middle line:90% Let's try that, 95252222. 00:06:16.305 --> 00:06:19.550 align:middle line:84% And let's go back to R1 before we hit Enter, 00:06:19.550 --> 00:06:23.000 align:middle line:84% and let's do some space so we can identify our messages. 00:06:23.000 --> 00:06:26.630 align:middle line:84% And let's go back here and hit the Call button. 00:06:26.630 --> 00:06:31.670 align:middle line:84% So this should not ring on the PSTN. 00:06:31.670 --> 00:06:35.240 align:middle line:84% If I look at the messages, I see here 00:06:35.240 --> 00:06:41.660 align:middle line:84% a 486, which you can see here a 486 busy message was sent back 00:06:41.660 --> 00:06:49.340 align:middle line:84% up to your SME, which is a 14, 64.14 IP address, which 00:06:49.340 --> 00:06:50.900 align:middle line:90% is the IP address of SME. 00:06:50.900 --> 00:06:53.360 align:middle line:84% So he sends-- the cube sends back a busy message, 00:06:53.360 --> 00:06:59.180 align:middle line:84% a busy SIP message 486 saying that I cannot drop this call 00:06:59.180 --> 00:07:03.050 align:middle line:84% because we've set the total amount of calls to just one. 00:07:03.050 --> 00:07:04.580 align:middle line:90% As soon as I-- 00:07:04.580 --> 00:07:07.940 align:middle line:84% as soon as HQ, which is HQ disconnects 00:07:07.940 --> 00:07:10.610 align:middle line:84% the call, now site B should be able to make 00:07:10.610 --> 00:07:13.800 align:middle line:84% the call because now it's not busy anymore. 00:07:13.800 --> 00:07:17.180 align:middle line:84% So you see, we can do a call treatment on the cube. 00:07:17.180 --> 00:07:20.685 align:middle line:84% You could do the same thing on call manager also, 00:07:20.685 --> 00:07:22.310 align:middle line:84% but because the question states that we 00:07:22.310 --> 00:07:24.447 align:middle line:84% need to do this on the cube, on the cube 00:07:24.447 --> 00:07:25.530 align:middle line:90% is pretty straightforward. 00:07:25.530 --> 00:07:27.650 align:middle line:84% All you have to do is enable call treatment, 00:07:27.650 --> 00:07:29.330 align:middle line:84% define what message you want to send, 00:07:29.330 --> 00:07:32.420 align:middle line:84% and define the how many amount of calls 00:07:32.420 --> 00:07:35.450 align:middle line:84% do you want to support simultaneously. 00:07:35.450 --> 00:07:37.310 align:middle line:84% I hope this video was informative for you, 00:07:37.310 --> 00:07:40.090 align:middle line:90% and thank you for watching. 00:07:40.090 --> 00:07:41.000 align:middle line:90%