WEBVTT 00:00:00.000 --> 00:00:03.451 align:middle line:90% 00:00:03.451 --> 00:00:06.902 align:middle line:90% [AUDIO LOGO] 00:00:06.902 --> 00:00:09.870 align:middle line:90% 00:00:09.870 --> 00:00:12.282 align:middle line:90% Hello, and welcome to INE. 00:00:12.282 --> 00:00:13.770 align:middle line:84% Today in this video, we will look 00:00:13.770 --> 00:00:16.230 align:middle line:90% at section 3, Unity Connection. 00:00:16.230 --> 00:00:21.480 align:middle line:84% Basically, in this task, we would configure our HQ site 00:00:21.480 --> 00:00:23.880 align:middle line:90% to get access to voicemail. 00:00:23.880 --> 00:00:26.490 align:middle line:84% So let's look at this task, task 3.1. 00:00:26.490 --> 00:00:28.200 align:middle line:84% Here in this task it says that we 00:00:28.200 --> 00:00:31.920 align:middle line:84% need to integrate the HQ UCM cluster and the Unity 00:00:31.920 --> 00:00:33.600 align:middle line:90% Connection using SIP. 00:00:33.600 --> 00:00:36.180 align:middle line:84% So basically, it's going to be a SIP integration. 00:00:36.180 --> 00:00:40.546 align:middle line:84% Set the voicemail pilot number to 2220. 00:00:40.546 --> 00:00:43.060 align:middle line:84% And after the integration is completed, 00:00:43.060 --> 00:00:46.470 align:middle line:84% you should be able to call into the voicemail pilot number 00:00:46.470 --> 00:00:48.750 align:middle line:90% from the user Tony Stark. 00:00:48.750 --> 00:00:51.480 align:middle line:84% Now, if you remember Tony Stark from the earlier sections 00:00:51.480 --> 00:00:55.540 align:middle line:84% was registered using Jabber 1 machine. 00:00:55.540 --> 00:00:57.990 align:middle line:84% So on Jabber 1 machine, Tony Stark 00:00:57.990 --> 00:01:00.660 align:middle line:84% is already preconfigured to log in. 00:01:00.660 --> 00:01:05.459 align:middle line:84% Now, make sure the local PSTN phone can also place a call 00:01:05.459 --> 00:01:11.460 align:middle line:90% from line 1 to 1-408-202-2220. 00:01:11.460 --> 00:01:13.920 align:middle line:84% And once he makes the call, he should 00:01:13.920 --> 00:01:17.490 align:middle line:90% hear the CUC system greeting. 00:01:17.490 --> 00:01:19.860 align:middle line:84% The call should be forwarded to voicemail 00:01:19.860 --> 00:01:23.250 align:middle line:84% if the HQ users are busy on a call 00:01:23.250 --> 00:01:25.920 align:middle line:84% or if they do not answer the call. 00:01:25.920 --> 00:01:28.050 align:middle line:84% If they don't answer the call, then the call 00:01:28.050 --> 00:01:31.500 align:middle line:84% should be forwarded to the voicemail in 20 seconds. 00:01:31.500 --> 00:01:34.470 align:middle line:84% The maximum amount of calls or the maximum number 00:01:34.470 --> 00:01:38.370 align:middle line:84% of calls on a line should be restricted to one. 00:01:38.370 --> 00:01:43.453 align:middle line:84% So let's quickly go and do this or configure this. 00:01:43.453 --> 00:01:45.120 align:middle line:84% So the first thing, what I'm going to do 00:01:45.120 --> 00:01:47.880 align:middle line:84% is I will configure the call manager side first, 00:01:47.880 --> 00:01:51.040 align:middle line:84% and then I will build the Unity Connection side. 00:01:51.040 --> 00:01:56.160 align:middle line:84% So on the call manager side, the first thing that I would do 00:01:56.160 --> 00:02:03.340 align:middle line:84% is I would go to System and go down to, 00:02:03.340 --> 00:02:06.940 align:middle line:84% I believe I would go down to us-- 00:02:06.940 --> 00:02:10.009 align:middle line:84% I would basically create a SIP Trunk Security Profile. 00:02:10.009 --> 00:02:11.740 align:middle line:90% So let's do that. 00:02:11.740 --> 00:02:18.860 align:middle line:90% 00:02:18.860 --> 00:02:25.880 align:middle line:84% So System, go-- I need to reduce the screen size a bit 00:02:25.880 --> 00:02:28.100 align:middle line:90% for it to fit in the screen. 00:02:28.100 --> 00:02:34.140 align:middle line:84% So go down to Security, go to SIP Trunk Security Profile. 00:02:34.140 --> 00:02:37.720 align:middle line:84% And do Find, copy the default one. 00:02:37.720 --> 00:02:43.600 align:middle line:84% So let's copy the Non Secure SIP Trunk Profile, 00:02:43.600 --> 00:02:47.260 align:middle line:90% and let's name this as CUC. 00:02:47.260 --> 00:02:48.610 align:middle line:90% And scroll down. 00:02:48.610 --> 00:02:51.710 align:middle line:84% And here we would need three checkboxes to be enabled. 00:02:51.710 --> 00:02:55.540 align:middle line:84% The first one is going to be accept out-of-dialog refer, 00:02:55.540 --> 00:03:00.590 align:middle line:84% accept unsolicited notification, and accept replaces header. 00:03:00.590 --> 00:03:02.980 align:middle line:90% Let's save this. 00:03:02.980 --> 00:03:06.850 align:middle line:84% The next thing that I would do is create a SIP Trunk 00:03:06.850 --> 00:03:09.250 align:middle line:90% to Unity Connection. 00:03:09.250 --> 00:03:10.910 align:middle line:90% So let's go and do that. 00:03:10.910 --> 00:03:14.660 align:middle line:90% So go down to Trunks. 00:03:14.660 --> 00:03:15.530 align:middle line:90% Let's do a find. 00:03:15.530 --> 00:03:17.820 align:middle line:84% I believe we only have two trunks right now. 00:03:17.820 --> 00:03:20.930 align:middle line:84% One is going to IM-P, and one is going 00:03:20.930 --> 00:03:23.250 align:middle line:90% to SME from the earlier task. 00:03:23.250 --> 00:03:26.240 align:middle line:84% So all we need to do is create a new SIP trunk which is going 00:03:26.240 --> 00:03:29.540 align:middle line:90% to point to Unity Connection. 00:03:29.540 --> 00:03:33.470 align:middle line:84% So let's go into trunk type as SIP. 00:03:33.470 --> 00:03:36.310 align:middle line:90% Let's do Next. 00:03:36.310 --> 00:03:39.176 align:middle line:84% Let's name this as SIP-TRUNK-TO-CUC. 00:03:39.176 --> 00:03:44.950 align:middle line:90% 00:03:44.950 --> 00:03:48.340 align:middle line:84% And this would be in the default device pool. 00:03:48.340 --> 00:03:49.180 align:middle line:90% Scroll down. 00:03:49.180 --> 00:03:51.850 align:middle line:84% Make sure you enable run on all active nodes, 00:03:51.850 --> 00:03:55.120 align:middle line:84% because in the lab exam, you would have a subscriber also. 00:03:55.120 --> 00:03:57.730 align:middle line:90% And scroll down. 00:03:57.730 --> 00:04:02.350 align:middle line:84% Enable redirecting diversion header for inbound, 00:04:02.350 --> 00:04:07.000 align:middle line:84% enable redirecting diversion header for outbound, 00:04:07.000 --> 00:04:11.620 align:middle line:84% and then give the FQDN or the IP address of your Unity 00:04:11.620 --> 00:04:12.310 align:middle line:90% Connection. 00:04:12.310 --> 00:04:19.120 align:middle line:84% Now, the IP address of Unity Connection is 11.100.64.15. 00:04:19.120 --> 00:04:20.920 align:middle line:84% However, if you do not know that, 00:04:20.920 --> 00:04:22.960 align:middle line:84% you can always go back to your DNS machine 00:04:22.960 --> 00:04:25.960 align:middle line:84% and check the A record that what's the FQDN address? 00:04:25.960 --> 00:04:28.070 align:middle line:90% And what's the IP address. 00:04:28.070 --> 00:04:31.000 align:middle line:84% So I'm going to give the FQDN here, so hq-cuc.ine.com. 00:04:31.000 --> 00:04:33.958 align:middle line:90% 00:04:33.958 --> 00:04:35.950 align:middle line:84% And SIP Trunk Security Profile is going 00:04:35.950 --> 00:04:37.720 align:middle line:90% to be the one that we created. 00:04:37.720 --> 00:04:41.710 align:middle line:84% And the SIP profile would be the default one. 00:04:41.710 --> 00:04:43.690 align:middle line:90% Let's save this. 00:04:43.690 --> 00:04:46.690 align:middle line:84% Now, once I save this, the first thing that I would do 00:04:46.690 --> 00:04:49.000 align:middle line:90% is I would reset my SIP Trunk. 00:04:49.000 --> 00:04:52.240 align:middle line:84% You must always reset your SIP Trunk when you create them 00:04:52.240 --> 00:04:55.520 align:middle line:90% or when you make any changes. 00:04:55.520 --> 00:04:57.685 align:middle line:84% So let's quickly reset my SIP trunk. 00:04:57.685 --> 00:05:00.820 align:middle line:90% 00:05:00.820 --> 00:05:06.490 align:middle line:84% The next thing that I would do is define a voicemail pilot 00:05:06.490 --> 00:05:08.470 align:middle line:90% number. 00:05:08.470 --> 00:05:10.290 align:middle line:84% So we are not doing skinny integration. 00:05:10.290 --> 00:05:13.260 align:middle line:84% So we don't need to go through the voicemail pilot wizard. 00:05:13.260 --> 00:05:16.290 align:middle line:84% We would basically create a voicemail pilot, 00:05:16.290 --> 00:05:18.640 align:middle line:84% and we would create a route pattern. 00:05:18.640 --> 00:05:20.640 align:middle line:84% So let's first create the voicemail pilot. 00:05:20.640 --> 00:05:27.030 align:middle line:84% So do a Find and choose the default voicemail pilot. 00:05:27.030 --> 00:05:30.038 align:middle line:84% So click on Default and give the pilot number. 00:05:30.038 --> 00:05:31.080 align:middle line:90% And ask for the question. 00:05:31.080 --> 00:05:34.050 align:middle line:84% The pilot number was supposed to be 2220. 00:05:34.050 --> 00:05:35.970 align:middle line:84% Make sure you enable this checkbox which 00:05:35.970 --> 00:05:39.400 align:middle line:84% says make this the default voicemail pilot for the system. 00:05:39.400 --> 00:05:41.310 align:middle line:90% Let's save this. 00:05:41.310 --> 00:05:44.760 align:middle line:84% Then we go down to, again, Advanced Features, Voicemail, 00:05:44.760 --> 00:05:47.310 align:middle line:90% and go to Voicemail Profile. 00:05:47.310 --> 00:05:52.270 align:middle line:84% And in the Voicemail Profile, we would choose the default one. 00:05:52.270 --> 00:05:56.340 align:middle line:84% And make sure that the voicemail pilot number is selected 00:05:56.340 --> 00:05:58.060 align:middle line:90% in the Voicemail Profile. 00:05:58.060 --> 00:06:01.440 align:middle line:84% So basically what we're doing is we create a voicemail pilot 00:06:01.440 --> 00:06:02.310 align:middle line:90% number. 00:06:02.310 --> 00:06:06.330 align:middle line:84% We attach the voicemail pilot number to the voicemail profile 00:06:06.330 --> 00:06:09.960 align:middle line:84% because the voicemail profile will get attached to the users 00:06:09.960 --> 00:06:11.530 align:middle line:90% under the line. 00:06:11.530 --> 00:06:15.020 align:middle line:84% So now let's go ahead and save this. 00:06:15.020 --> 00:06:17.980 align:middle line:84% So make sure you have 2220 selected. 00:06:17.980 --> 00:06:21.460 align:middle line:84% And once you have done that, we will now go and create 00:06:21.460 --> 00:06:28.600 align:middle line:84% a route pattern which will be for 2220 going 00:06:28.600 --> 00:06:31.820 align:middle line:84% to the SIP Trunk, which is Unity Connection that we created. 00:06:31.820 --> 00:06:37.880 align:middle line:84% So let's create a route pattern which is going to be 2220. 00:06:37.880 --> 00:06:40.580 align:middle line:84% We don't need to put him in any partition 00:06:40.580 --> 00:06:44.240 align:middle line:84% or give him access to any CSCs because my lines 00:06:44.240 --> 00:06:47.670 align:middle line:84% are in the default partition, which is non partition. 00:06:47.670 --> 00:06:50.960 align:middle line:84% So I don't really need to give my pilot number 00:06:50.960 --> 00:06:54.290 align:middle line:84% or my voicemail profile any access to any access CSCs 00:06:54.290 --> 00:06:56.240 align:middle line:84% because they are in the non partition. 00:06:56.240 --> 00:06:58.230 align:middle line:84% My lines are in the non partition. 00:06:58.230 --> 00:07:00.530 align:middle line:90% So let's do 2220. 00:07:00.530 --> 00:07:01.550 align:middle line:90% Let's dismiss this. 00:07:01.550 --> 00:07:05.900 align:middle line:84% Scroll down to gateway is going to be the Unity Connection. 00:07:05.900 --> 00:07:08.600 align:middle line:84% Let's uncheck the outside dial tone 00:07:08.600 --> 00:07:12.710 align:middle line:84% and enable the urgent priority and save the settings. 00:07:12.710 --> 00:07:14.960 align:middle line:90% Let's save this. 00:07:14.960 --> 00:07:18.230 align:middle line:84% Now once we've saved this, the next thing that I would do 00:07:18.230 --> 00:07:22.100 align:middle line:84% is go to my phones and ensure that I 00:07:22.100 --> 00:07:24.943 align:middle line:84% have all the checkboxes enabled as per the requirement 00:07:24.943 --> 00:07:25.610 align:middle line:90% of the question. 00:07:25.610 --> 00:07:28.010 align:middle line:84% So the first thing I need to do is, under the line, 00:07:28.010 --> 00:07:31.940 align:middle line:84% I need to enable the checkbox for voicemail profile 00:07:31.940 --> 00:07:34.640 align:middle line:84% because the pilot number is inside the voicemail profile, 00:07:34.640 --> 00:07:37.340 align:middle line:84% and the voicemail profile has to be attached to the line. 00:07:37.340 --> 00:07:39.020 align:middle line:90% And then I need to do-- 00:07:39.020 --> 00:07:43.250 align:middle line:84% if he's busy or if he does not answer or if he's unregistered, 00:07:43.250 --> 00:07:45.230 align:middle line:90% I should forward to voicemail. 00:07:45.230 --> 00:07:47.390 align:middle line:84% My busy trigger should be set to 1, 00:07:47.390 --> 00:07:51.110 align:middle line:84% and no answer time limit should be set to 20 seconds. 00:07:51.110 --> 00:07:55.610 align:middle line:84% Now, even though the question states about only about testing 00:07:55.610 --> 00:07:58.160 align:middle line:84% from Tony Stark, what I would recommend in the lab exam, 00:07:58.160 --> 00:07:59.990 align:middle line:84% if you have two users or three users, 00:07:59.990 --> 00:08:02.550 align:middle line:84% do it for all the users for all your lines. 00:08:02.550 --> 00:08:05.540 align:middle line:84% So what I'm going to do is go through Bulk Administration, 00:08:05.540 --> 00:08:09.860 align:middle line:84% go to Phones, go to Add Update Lines, 00:08:09.860 --> 00:08:12.450 align:middle line:90% and click on Update Lines. 00:08:12.450 --> 00:08:17.350 align:middle line:84% So let's go back here, go to Update Lines and attach. 00:08:17.350 --> 00:08:18.400 align:middle line:90% Let's do a Find. 00:08:18.400 --> 00:08:20.860 align:middle line:84% We should find three directory numbers here-- 00:08:20.860 --> 00:08:26.170 align:middle line:84% 2001 which is HQ one, 2002 which is HQ four and two, and 2003 00:08:26.170 --> 00:08:27.310 align:middle line:90% which is Tony Stark. 00:08:27.310 --> 00:08:29.820 align:middle line:90% Let's do a Next. 00:08:29.820 --> 00:08:33.270 align:middle line:84% The first thing that I would do is scroll down a bit. 00:08:33.270 --> 00:08:35.640 align:middle line:84% Enable the checkbox for voicemail profile 00:08:35.640 --> 00:08:38.520 align:middle line:84% and select the default voicemail profile which contains 00:08:38.520 --> 00:08:40.620 align:middle line:90% the voicemail pilot number. 00:08:40.620 --> 00:08:43.299 align:middle line:90% Scroll down. 00:08:43.299 --> 00:08:49.210 align:middle line:84% And forward, if he's busy, then forward to voicemail. 00:08:49.210 --> 00:08:53.890 align:middle line:84% But so far we're busy internal, forward busy external, 00:08:53.890 --> 00:08:55.330 align:middle line:90% forward busy-- 00:08:55.330 --> 00:08:59.080 align:middle line:84% oh, forward no answer internal to voicemail, and forward no 00:08:59.080 --> 00:09:01.470 align:middle line:90% answer external. 00:09:01.470 --> 00:09:06.930 align:middle line:84% Then scroll down to forward unregistered internal, forward 00:09:06.930 --> 00:09:08.970 align:middle line:90% unregistered external. 00:09:08.970 --> 00:09:12.930 align:middle line:84% No answer duration should be set to 20 seconds. 00:09:12.930 --> 00:09:17.330 align:middle line:84% And then busy trigger which is going to be at the bottom, 00:09:17.330 --> 00:09:19.750 align:middle line:84% so busy trigger is going to be set to 1. 00:09:19.750 --> 00:09:23.380 align:middle line:84% So if he's on a phone call, then the next phone, 00:09:23.380 --> 00:09:27.190 align:middle line:84% next call that comes in would go to the voicemail directly. 00:09:27.190 --> 00:09:32.270 align:middle line:84% Let's go and say run immediately and Submit. 00:09:32.270 --> 00:09:38.050 align:middle line:84% So now, all my phones should have access to-- 00:09:38.050 --> 00:09:40.610 align:middle line:84% it should have the access to the voicemail pilot number based 00:09:40.610 --> 00:09:42.610 align:middle line:84% on their lines because the voicemail profile has 00:09:42.610 --> 00:09:43.330 align:middle line:90% been attached. 00:09:43.330 --> 00:09:46.510 align:middle line:84% The voicemail profile is containing the voicemail pilot 00:09:46.510 --> 00:09:47.110 align:middle line:90% number. 00:09:47.110 --> 00:09:49.810 align:middle line:84% And the pilot number I have a route pattern 00:09:49.810 --> 00:09:52.030 align:middle line:84% that matches the pilot number that 00:09:52.030 --> 00:09:55.030 align:middle line:84% points to the SIP Trunk going to Unity Connection. 00:09:55.030 --> 00:09:56.980 align:middle line:84% The next thing that I would need to do 00:09:56.980 --> 00:09:59.350 align:middle line:84% is I would need to modify my Jabber. 00:09:59.350 --> 00:10:03.280 align:middle line:84% Because in the lab exam, again, you do not basically 00:10:03.280 --> 00:10:04.270 align:middle line:90% have physical phones. 00:10:04.270 --> 00:10:05.812 align:middle line:84% There are no physical phones anymore. 00:10:05.812 --> 00:10:07.600 align:middle line:84% You only have Jabber, which means 00:10:07.600 --> 00:10:09.190 align:middle line:84% that for you to test your voicemail, 00:10:09.190 --> 00:10:12.220 align:middle line:84% you would need voicemail modifier work on Jabber. 00:10:12.220 --> 00:10:16.122 align:middle line:84% So what I'm going to do is go to-- 00:10:16.122 --> 00:10:17.080 align:middle line:90% in fact, you know what? 00:10:17.080 --> 00:10:19.030 align:middle line:84% For this task, I don't really need 00:10:19.030 --> 00:10:21.590 align:middle line:84% to do that because as of right now, 00:10:21.590 --> 00:10:25.960 align:middle line:84% I only need to test calling the Unity Connection 00:10:25.960 --> 00:10:27.970 align:middle line:90% and during the system greeting. 00:10:27.970 --> 00:10:30.130 align:middle line:90% I don't really have to do-- 00:10:30.130 --> 00:10:34.390 align:middle line:84% check my voicemail or check my messages, basically. 00:10:34.390 --> 00:10:37.400 align:middle line:84% I don't have to do that in this task at least. 00:10:37.400 --> 00:10:40.210 align:middle line:84% So again, in this task, I don't really 00:10:40.210 --> 00:10:44.860 align:middle line:84% need to go and go and modify my service profile and my UC 00:10:44.860 --> 00:10:49.360 align:middle line:84% service because I don't have to go and leave 00:10:49.360 --> 00:10:53.470 align:middle line:84% a voicemail or a message on somebody's extension. 00:10:53.470 --> 00:10:57.310 align:middle line:84% So I believe everything that we needed 00:10:57.310 --> 00:11:00.130 align:middle line:84% is done on the call manager side. 00:11:00.130 --> 00:11:03.250 align:middle line:84% Let's now go to Unity Connection. 00:11:03.250 --> 00:11:05.960 align:middle line:84% It's asking me to-- my passport has expired. 00:11:05.960 --> 00:11:10.282 align:middle line:84% So let's regive the password as c1sc0123! 00:11:10.282 --> 00:11:11.820 align:middle line:90% And let's say Save. 00:11:11.820 --> 00:11:16.990 align:middle line:90% 00:11:16.990 --> 00:11:19.720 align:middle line:84% And here, the first thing that we would do 00:11:19.720 --> 00:11:22.250 align:middle line:90% is go to the Phone System. 00:11:22.250 --> 00:11:24.040 align:middle line:90% So go back down to Phone System. 00:11:24.040 --> 00:11:35.590 align:middle line:90% 00:11:35.590 --> 00:11:37.960 align:middle line:84% You would see the default phone system here. 00:11:37.960 --> 00:11:40.660 align:middle line:84% You can use the default phone system for HQ, 00:11:40.660 --> 00:11:44.080 align:middle line:84% or you could create your own phone system for HQ. 00:11:44.080 --> 00:11:46.400 align:middle line:84% By default, there's a built-in phone system, 00:11:46.400 --> 00:11:47.260 align:middle line:90% which you can use. 00:11:47.260 --> 00:11:49.210 align:middle line:90% You can modify that if you want. 00:11:49.210 --> 00:11:52.430 align:middle line:84% But again, if you want to create your own phone system, 00:11:52.430 --> 00:11:53.780 align:middle line:90% you can do that as well. 00:11:53.780 --> 00:11:56.485 align:middle line:84% So there's no restriction as such in the lab exam. 00:11:56.485 --> 00:11:59.310 align:middle line:90% 00:11:59.310 --> 00:12:02.040 align:middle line:90% So let's click on Phone System. 00:12:02.040 --> 00:12:03.740 align:middle line:84% I'll use the default phone system. 00:12:03.740 --> 00:12:09.420 align:middle line:84% So click on Phone System, and let's change this name to HQ, 00:12:09.420 --> 00:12:13.100 align:middle line:90% and let's do Save. 00:12:13.100 --> 00:12:14.960 align:middle line:90% Next thing I would go to Edit. 00:12:14.960 --> 00:12:18.000 align:middle line:90% 00:12:18.000 --> 00:12:21.960 align:middle line:84% I don't need to specify the AXL servers at this point of time 00:12:21.960 --> 00:12:25.350 align:middle line:84% in this question because I'm not really importing any users 00:12:25.350 --> 00:12:26.590 align:middle line:90% at this point of time. 00:12:26.590 --> 00:12:28.890 align:middle line:84% So I don't really need to do because I haven't really 00:12:28.890 --> 00:12:31.500 align:middle line:84% read the next question, which will probably 00:12:31.500 --> 00:12:33.930 align:middle line:84% be that I need to create the users manually 00:12:33.930 --> 00:12:36.220 align:middle line:90% or maybe import the users. 00:12:36.220 --> 00:12:37.830 align:middle line:84% So if I need to import the users, 00:12:37.830 --> 00:12:40.780 align:middle line:84% then I would go and add the AXL server information, 00:12:40.780 --> 00:12:43.320 align:middle line:84% which is going to be a publisher and subscriber. 00:12:43.320 --> 00:12:45.690 align:middle line:84% However, as for this question, I don't really 00:12:45.690 --> 00:12:46.750 align:middle line:90% have to do anything. 00:12:46.750 --> 00:12:48.510 align:middle line:84% All I do is create the phone system. 00:12:48.510 --> 00:12:52.170 align:middle line:84% Then go to Port Groups, create a new port group 00:12:52.170 --> 00:12:54.010 align:middle line:90% that uses that phone system. 00:12:54.010 --> 00:12:56.810 align:middle line:84% So this would change the phone system as HQ. 00:12:56.810 --> 00:13:01.760 align:middle line:84% This would change to SIP, and you can leave the display name 00:13:01.760 --> 00:13:02.750 align:middle line:90% as the default. 00:13:02.750 --> 00:13:04.880 align:middle line:84% Scroll down, and here you would give 00:13:04.880 --> 00:13:08.300 align:middle line:84% the IP address of your subscriber in the lab exam. 00:13:08.300 --> 00:13:11.450 align:middle line:84% And then you would go and add the publisher also. 00:13:11.450 --> 00:13:14.060 align:middle line:84% In our case, we only have one publisher. 00:13:14.060 --> 00:13:19.730 align:middle line:84% We don't have subscribers, so I would just say hq-cucm.ine.com. 00:13:19.730 --> 00:13:21.890 align:middle line:84% You can give the IP address, also. 00:13:21.890 --> 00:13:22.880 align:middle line:90% Let's save this. 00:13:22.880 --> 00:13:25.340 align:middle line:84% But let's say in the lab exam, you had subscriber. 00:13:25.340 --> 00:13:28.130 align:middle line:84% Also, in that case, you would give the subscriber's IP 00:13:28.130 --> 00:13:32.240 align:middle line:84% address here and then go to Edit, go to Servers. 00:13:32.240 --> 00:13:36.440 align:middle line:84% And here you would add your publisher also 00:13:36.440 --> 00:13:40.910 align:middle line:84% for SIP servers and TFTP servers. 00:13:40.910 --> 00:13:43.640 align:middle line:84% Right now, I only have one publisher, 00:13:43.640 --> 00:13:45.170 align:middle line:90% so I'm just adding one. 00:13:45.170 --> 00:13:46.850 align:middle line:84% I don't have to go back to servers. 00:13:46.850 --> 00:13:50.270 align:middle line:84% Let's go down to Codec Advertising. 00:13:50.270 --> 00:13:53.180 align:middle line:84% Now, by default, your Unity Connection 00:13:53.180 --> 00:13:56.705 align:middle line:90% supports G.711 and G.729. 00:13:56.705 --> 00:13:58.580 align:middle line:84% Now if you look at the question, the question 00:13:58.580 --> 00:14:03.230 align:middle line:84% says HQ users are going to dial the pilot number, 00:14:03.230 --> 00:14:05.060 align:middle line:84% and they should hear the system greeting. 00:14:05.060 --> 00:14:08.870 align:middle line:84% And even your PSTN is going to dial. 00:14:08.870 --> 00:14:11.060 align:middle line:84% Now, if you remember from the dial plan 00:14:11.060 --> 00:14:14.630 align:middle line:84% that we had done in section 2 earlier, 00:14:14.630 --> 00:14:19.160 align:middle line:84% PSTN was calling to the-- calling and going to the cube. 00:14:19.160 --> 00:14:20.870 align:middle line:90% Cube was sending it to SME. 00:14:20.870 --> 00:14:24.090 align:middle line:84% SME was sending it to HQ, and HQ would forward that to Unity 00:14:24.090 --> 00:14:25.220 align:middle line:90% Connection. 00:14:25.220 --> 00:14:28.160 align:middle line:84% The thing is that if your dial peers do not 00:14:28.160 --> 00:14:31.460 align:middle line:84% have a codec pass through, pass through content SDP, which 00:14:31.460 --> 00:14:35.600 align:middle line:84% means your dial peers are going to change the SDP information, 00:14:35.600 --> 00:14:38.510 align:middle line:84% change your codec information, which 00:14:38.510 --> 00:14:40.700 align:middle line:90% could require a transcoder. 00:14:40.700 --> 00:14:44.090 align:middle line:84% But because we have configured-- if you look at my dial peers 00:14:44.090 --> 00:14:48.020 align:middle line:84% on, let's say, R1, if you look at my inbound dial peer, 00:14:48.020 --> 00:14:49.790 align:middle line:90% my outbound dial peers-- 00:14:49.790 --> 00:14:54.320 align:middle line:84% let's do a show run section dial-peer 2000 voip, 00:14:54.320 --> 00:14:56.810 align:middle line:84% this is my inbound dial coming from PSTN. 00:14:56.810 --> 00:14:59.810 align:middle line:90% I have content pass through SDP. 00:14:59.810 --> 00:15:02.630 align:middle line:84% And if I look at my outbound dial peer, which 00:15:02.630 --> 00:15:06.410 align:middle line:84% is, I believe it's 30, which is going to SME, 00:15:06.410 --> 00:15:08.875 align:middle line:84% again, that has pass through content SDP, which 00:15:08.875 --> 00:15:10.250 align:middle line:84% means my dial peers are not going 00:15:10.250 --> 00:15:14.600 align:middle line:84% to modify any SDP information or codec information. 00:15:14.600 --> 00:15:16.640 align:middle line:84% They would just pass through that information. 00:15:16.640 --> 00:15:18.840 align:middle line:84% Because the thing is that, in the lab exam, 00:15:18.840 --> 00:15:21.980 align:middle line:84% there is no transcoder support, because we're 00:15:21.980 --> 00:15:24.680 align:middle line:84% using virtual routers, which means transcoder 00:15:24.680 --> 00:15:25.830 align:middle line:90% is not going to work. 00:15:25.830 --> 00:15:29.090 align:middle line:84% So make sure that you do not invoke 00:15:29.090 --> 00:15:31.010 align:middle line:84% or try to invoke a transcoder because you 00:15:31.010 --> 00:15:32.400 align:middle line:90% don't have the resources. 00:15:32.400 --> 00:15:36.260 align:middle line:84% So what I would suggest is in the lab exam, 00:15:36.260 --> 00:15:38.870 align:middle line:90% go and just use G.711. 00:15:38.870 --> 00:15:40.370 align:middle line:90% Go and remove G.729. 00:15:40.370 --> 00:15:42.140 align:middle line:84% So everywhere we are using G.711, 00:15:42.140 --> 00:15:43.640 align:middle line:84% so we don't really have a problem. 00:15:43.640 --> 00:15:44.645 align:middle line:90% Let's save this. 00:15:44.645 --> 00:15:47.230 align:middle line:90% 00:15:47.230 --> 00:15:51.090 align:middle line:84% And next, I would go and create ports. 00:15:51.090 --> 00:15:53.540 align:middle line:84% So of ports, how many ports can I create? 00:15:53.540 --> 00:15:56.880 align:middle line:84% Now, based on your licensing, now usually in the lab exam, 00:15:56.880 --> 00:15:59.340 align:middle line:84% they would tell you how many ports are required. 00:15:59.340 --> 00:16:02.863 align:middle line:84% So maybe your port numbers are going to be-- 00:16:02.863 --> 00:16:04.280 align:middle line:84% if it's a skinny integration, they 00:16:04.280 --> 00:16:07.910 align:middle line:84% would tell you the port numbers and the pilot number also. 00:16:07.910 --> 00:16:10.190 align:middle line:84% If it's a SIP integration, they don't really 00:16:10.190 --> 00:16:13.280 align:middle line:84% mention anything about how many ports are required. 00:16:13.280 --> 00:16:14.980 align:middle line:84% It's going to be based on the licenses. 00:16:14.980 --> 00:16:16.610 align:middle line:84% So what I would suggest is, maybe 00:16:16.610 --> 00:16:19.010 align:middle line:90% create two or three ports. 00:16:19.010 --> 00:16:22.700 align:middle line:84% In my case, because we're using the evaluation license, 00:16:22.700 --> 00:16:25.740 align:middle line:90% so it only supports two ports. 00:16:25.740 --> 00:16:30.320 align:middle line:84% So what I'm going to do is I am not going to use both my ports. 00:16:30.320 --> 00:16:32.820 align:middle line:90% I would just use one port. 00:16:32.820 --> 00:16:35.540 align:middle line:84% So let's just choose one port and save. 00:16:35.540 --> 00:16:37.040 align:middle line:90% I'm not going to use two ports. 00:16:37.040 --> 00:16:41.010 align:middle line:84% It won't let me create more than two ports because of licensing. 00:16:41.010 --> 00:16:43.100 align:middle line:84% So if I try and give three, it would only 00:16:43.100 --> 00:16:45.210 align:middle line:90% create two ports and not three. 00:16:45.210 --> 00:16:47.930 align:middle line:84% But what I'm doing right now is I'm not really 00:16:47.930 --> 00:16:49.100 align:middle line:90% using both my ports. 00:16:49.100 --> 00:16:52.950 align:middle line:84% I'm just using one port because maybe in the next question 00:16:52.950 --> 00:16:56.240 align:middle line:84% I may have to do a skinny integration with site B. 00:16:56.240 --> 00:16:57.740 align:middle line:90% So I don't really know-- 00:16:57.740 --> 00:17:01.310 align:middle line:84% so don't use all your ports for just one side, 00:17:01.310 --> 00:17:04.779 align:middle line:84% because if you're using the same Unity Connection for both 00:17:04.779 --> 00:17:07.910 align:middle line:84% your sites, HQ and site B, you'll 00:17:07.910 --> 00:17:10.800 align:middle line:84% need to reserve some for site B also. 00:17:10.800 --> 00:17:15.650 align:middle line:84% So I would say because we only testing it from Tony Stark, 00:17:15.650 --> 00:17:17.270 align:middle line:90% one is more than enough. 00:17:17.270 --> 00:17:19.910 align:middle line:84% Or if you want, you can do two in the lab exam 00:17:19.910 --> 00:17:24.510 align:middle line:84% and maybe keep some free for site B. In our case, 00:17:24.510 --> 00:17:27.030 align:middle line:84% I'm just going to use one because my maximum support is 00:17:27.030 --> 00:17:27.530 align:middle line:90% two. 00:17:27.530 --> 00:17:30.350 align:middle line:84% So I'm going to keep one for HQ and one for site B 00:17:30.350 --> 00:17:34.850 align:middle line:84% If I have a task for site B. So just one port 00:17:34.850 --> 00:17:36.990 align:middle line:84% and I think that should be enough. 00:17:36.990 --> 00:17:42.740 align:middle line:84% Let's go to R1 and do a debug ccsip messages 00:17:42.740 --> 00:17:44.960 align:middle line:90% and go to my PSTN. 00:17:44.960 --> 00:17:51.737 align:middle line:84% Let's go to PSTN, and let's make a call. 00:17:51.737 --> 00:17:53.570 align:middle line:84% So as per the question, I'm supposed to call 00:17:53.570 --> 00:18:01.010 align:middle line:84% 1-408-202-2220, and it should be from line 1. 00:18:01.010 --> 00:18:04.580 align:middle line:84% Let's call this number and check that 00:18:04.580 --> 00:18:09.080 align:middle line:84% does it hit the route pattern or in fact the dial appears 00:18:09.080 --> 00:18:10.340 align:middle line:90% and forwards to SME? 00:18:10.340 --> 00:18:12.140 align:middle line:90% SME would forward to HQ. 00:18:12.140 --> 00:18:15.270 align:middle line:84% HQ would redirect that to Unity Connection. 00:18:15.270 --> 00:18:19.700 align:middle line:84% So I should be able to hear system greeting. 00:18:19.700 --> 00:18:24.540 align:middle line:84% Let's go back to Jabber 3 and see. 00:18:24.540 --> 00:18:29.150 align:middle line:84% So it is going to 2220, which you can see here that it 00:18:29.150 --> 00:18:31.110 align:middle line:90% is going to Unity Connection. 00:18:31.110 --> 00:18:33.680 align:middle line:84% And if I do a Control-Shift-S, now, 00:18:33.680 --> 00:18:37.438 align:middle line:84% because we don't have audio or sound, sound card, 00:18:37.438 --> 00:18:39.230 align:middle line:84% you cannot really hear the system greeting. 00:18:39.230 --> 00:18:42.230 align:middle line:84% But what you can do is do a Control-Shift-S, 00:18:42.230 --> 00:18:44.990 align:middle line:84% and you should see G.711 being invoked 00:18:44.990 --> 00:18:49.680 align:middle line:84% and the active call to 2220 which you can see here. 00:18:49.680 --> 00:18:52.880 align:middle line:84% So it did go to my Unity Connection. 00:18:52.880 --> 00:18:54.110 align:middle line:90% Let's close this. 00:18:54.110 --> 00:19:01.810 align:middle line:84% And the call go to Jabber 1 machine, go to Tony Stark, 00:19:01.810 --> 00:19:05.320 align:middle line:90% and let's call 2220 here. 00:19:05.320 --> 00:19:07.630 align:middle line:90% And let's make this call. 00:19:07.630 --> 00:19:09.010 align:middle line:90% It should go to the voicemail. 00:19:09.010 --> 00:19:10.150 align:middle line:90% It should connect. 00:19:10.150 --> 00:19:13.120 align:middle line:84% So if you see here, the timer has started. 00:19:13.120 --> 00:19:15.340 align:middle line:84% And if I do a Control-Shift-S, you 00:19:15.340 --> 00:19:18.820 align:middle line:84% can see that the call is active, and the codec 00:19:18.820 --> 00:19:23.020 align:middle line:84% is G.711 because we removed G.729 00:19:23.020 --> 00:19:26.170 align:middle line:84% from the list of codec advertisement 00:19:26.170 --> 00:19:27.370 align:middle line:90% for Unity Connection. 00:19:27.370 --> 00:19:30.730 align:middle line:84% So it only supports G.711 now, and it 00:19:30.730 --> 00:19:32.890 align:middle line:84% shouldn't be a problem anymore because I 00:19:32.890 --> 00:19:34.860 align:middle line:90% don't need a transcoder. 00:19:34.860 --> 00:19:37.380 align:middle line:84% So my task is basically complete. 00:19:37.380 --> 00:19:42.930 align:middle line:84% If I look back, let's say, if I go back to PSTN and I call Tony 00:19:42.930 --> 00:19:49.690 align:middle line:84% Stark now, 1-408-202-2003 and I don't answer the call in 20 00:19:49.690 --> 00:19:54.600 align:middle line:84% seconds, the call should get redirected to the voicemail. 00:19:54.600 --> 00:19:56.760 align:middle line:84% Would I be able to leave a voicemail? 00:19:56.760 --> 00:19:58.390 align:middle line:84% No, because at this point of time, 00:19:58.390 --> 00:20:02.190 align:middle line:84% we have not associated the mailbox with the user 00:20:02.190 --> 00:20:05.870 align:middle line:84% because I don't really have any users on Utility Connection. 00:20:05.870 --> 00:20:10.860 align:middle line:84% So I would not be able to leave a voicemail message, 00:20:10.860 --> 00:20:12.970 align:middle line:84% but I should hear the system greeting. 00:20:12.970 --> 00:20:16.020 align:middle line:84% Now, if you see, the call should have been redirected, 00:20:16.020 --> 00:20:19.050 align:middle line:84% which you can see here, it went to 2220. 00:20:19.050 --> 00:20:23.880 align:middle line:84% It got redirected to Unity Connection. 00:20:23.880 --> 00:20:25.410 align:middle line:90% Let's end this call. 00:20:25.410 --> 00:20:29.160 align:middle line:84% And with this, I believe we have successfully 00:20:29.160 --> 00:20:35.170 align:middle line:84% configured all the requirements of the task 00:20:35.170 --> 00:20:39.960 align:middle line:84% where we configured the end users, the lines 00:20:39.960 --> 00:20:42.000 align:middle line:90% with the voicemail profile. 00:20:42.000 --> 00:20:45.930 align:middle line:84% We did the call forward busy, call forward no answer, 00:20:45.930 --> 00:20:52.050 align:middle line:84% call forward unregistered, the no answer time to 20 seconds, 00:20:52.050 --> 00:20:55.560 align:middle line:90% and the busy trigger set to 1. 00:20:55.560 --> 00:20:58.000 align:middle line:84% With this, we've met all of the requirements. 00:20:58.000 --> 00:20:59.850 align:middle line:84% I hope this video was informative for you, 00:20:59.850 --> 00:21:02.500 align:middle line:90% and thank you for watching. 00:21:02.500 --> 00:21:03.000 align:middle line:90%