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So, now, we’re gonna get into the core of actually setting up a CUCME server
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and this time we’re gonna start with SIP CUCME server set up.
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We’re told to provision the Branch 2 Router 3 as a SIP CME server
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with the following stipulations:
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We’re told to allow maximum of two SIP IP phones
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and to ensure they upgrade as needed.
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To ensure that these phones can authenticate and register.
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To allow a maximum of 10 directory numbers.
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To provision the SIP server IP address.
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To be the same as the loop back 0 interface.
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and ensure that SIP packets from the router only come from that address.
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To ensure that the phones display their time and date using the formats below.
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Time, they should display 24-hour time in the same time zone as the router
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and get their time from, via NTP from the local router
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and the date should be in the format, day, month and then 2-digit year.
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Let’s switch back over to our router configuration and clear off our screen
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and we know that we have the ability to serve NTP
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because we’ve already looked at show NTP associations.
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I should say, do show NTP associations.
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And we are sinking to the PSTN but we also are an NTP master
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at a lower stratum so we can serve NTP.
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Let’s get in to the beginning of the SIP CME server
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which is done with the voice register global command.
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Voice register global is what turns on SIP CME.
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Now, the first thing we need to do is configure the mode that we’re gonna use.
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There’s a lot of information that we can do here.
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Actually, there’s not nearly as much as what we can do in telephony service.
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But the very first thing is we need to find the mode.
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Is it CME or SRST?
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and as we showed before, do show voice register global.
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The base mode is SRST.
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If we do show Run, Type 2 section voice register,
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we see that we've got voice register global, max DN of 144 and 42,
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those are just the defaults that it configured
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and it’s mode SRST. So, we actually have to say mode CME.
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Actually I'm going to just do.. Oops, that's not what I wanna do.
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Here we go. Let’s bring this down just a little bit.
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just in case anyone has the bottom of the screen cut off.
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So, we’re gonna do mode CME
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and now we see that the mode has been changed to CME version 701.
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0:03:29
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So, we need to define the max number of phones
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and we remember that those were called pools.
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So the max number of pools we were told should be set to 2
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and we were told the max number of DN should be 10.
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Let’s just go ahead and do that real quick since we’re here.
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Max DN of 10.
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So, now, if we do show voice register global, it’s changed to 2 and 10.
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And we were told to ensure that the phones can upgrade as necessary.
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That’s done with the load command.
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So we’ll say the load for the 7961 is the only phone that we have
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and we’re told that firmware file name without dot loads.
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So, let’s actually do, do show, sorry not run,
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do show flash type 2 include 7961
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and we've got terms 61 default loads, term 41 default loads
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and this SIP 418-4-2s.loads.
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So, we’re gonna use this one.
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So, all we wanna say is load 7961 and this file name without dot loads
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because it automatically depends on that.
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And we’re told to ensure that the phones can authenticate and register.
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So, we actually need to say Authenticate
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for we should tell the request to register that they have to authenticate.
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SIP needs to authenticate.
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We’ve already done the max DN.
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We also have to define the source interface
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as the same IP address as the loopback 0 which we remember
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do show IP interface 3 type 2 include loopback.
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It is 177.1.254.3, copy and paste is always a good idea if you can.
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So the source address is the command we use.
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We’ll paste in the address and we can put into the port if we want
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or just hit enter and this will automatically populate
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with port 5060, the default for SIP.
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Next, we were told, but actually first of all, that’s only half of this task
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because what we also need to do is jump out,
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go into voice service VOIP global,
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go into SIP and say bind and we can either bind media control
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or if we just type all, it’ll actually do both for us.
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Bind all communication for SIP signalling under voice service VOIP.
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so this global for the box, for the VOIP types of dial-peers
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which are going to be whether created for SIP phones
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and we’re gonna bind them to the interface of loopback 0.
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So, do show run type 2 section voice service.
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0:07:01
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Actually, I shouldn’t say Section, I should probably say "Begins with voice service."
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And it didn’t take for some reason, source interface,
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sorry not the source interface, bind all to the source interface of loopback 0.
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0:07:31
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Let’s just do them individually.
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0:07:36
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not !, bind, control to source interface, loopback 0,
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and bind media to the source interface of loopback 0.
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0:08:03
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Let’s go ahead while we’re here,
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I’m not sure why that’s not taking right now.
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We’ll investigate that in a minute.
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Let’s also turn on the SIP register server.
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0:08:11
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Now, we were told earlier when we stated,
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0:08:17
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I can scroll back up here,
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under voice registrar global to authenticate register,
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all that was saying is registration attempt should be authenticated.
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0:08:27
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The voice register global is essentially the
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0:08:30
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configuration parameters for the SIP server
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0:08:33
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but the router itself under voice service VOIP in subsection SIP
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is where we actually turn on the register server.
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0:08:42
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So, now, we have registered our server, we could even give it and expires
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0:08:46
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such as minimum or maximum of, up to 3600 minutes or an hour.
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It goes as low as 60 seconds or every minute
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and then a maximum value of up to,
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0:09:05
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I think that’s a day but we could say,
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let’s just say the minimum of the maximum,
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the lowest possible number for the maximum, I should better say.
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0:09:13
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Let’s try that bind command again after we did that,
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bind all to source interface loopback 0,
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do show run,
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it’s still not doing that.
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0:09:30
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Let’s exit out 1 and let’s say allow connection SIP to SIP.
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0:09:35
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We said we would need that.
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Let’s make sure that shows up.
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There we go, now let’s jump back in the subsection SIP,
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bind all to source loopback 0,
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0:09:59
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still not doing it.
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You know what? I know why we can’t do that,
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because we don’t actually have any dial-peers configured.
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We have to have at least one dial-peer voice.
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Let’s call it 10 type VOIP.
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0:10:15
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Now, let’s go back to voice service VOIP and back to SIP
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and now let’s say bind all to source loopback 0.
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There we go.
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0:10:33
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We have to have a dial-peer of the type VOIP
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0:10:35
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in order to actually have the possibility
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0:10:38
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that we would need to bind anything.
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Also, when I've created the voice register pool or voice register DN,
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that would also allow me to do it.
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In fact, just to illustrate that, let’s just go ahead
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and go back to we have create a dial peer 10.
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Let’s just uncreate that or delete that
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and go back to do show run type 2 section voice service.
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It actually stays in the configuration there.
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If I took them out and try to put them back in, it wouldn’t, that’s why.
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0:11:09
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So, once I had created the voice register pool and DN,
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0:11:12
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that would create a VOIP SIP dial-peer and then those commands would stick.
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0:11:19
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So, now going back to the task,
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0:11:27
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we were told to ensure the phones
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0:11:31
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displayed their time and date in the proper formats.
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0:11:35
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So back under voice register global,
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0:11:39
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we are going to look at,
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0:11:41
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now we can see there’s a lot more information
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0:11:43
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since we changed it to mode CME.
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0:11:47
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Now, we’re gonna look at our gate format and also our time format.
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0:12:01
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So, let’s just say time format to begin with.
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0:12:05
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Let’s say time zone to begin with.
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0:12:08
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Time zone, if we take a look to a question mark,
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we have a lot of options here
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and we can actually see that we have
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type 23, I’m sorry, type 24. Where is it?
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There you go.
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0:12:27
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Type 28 rather, essentially European standard and daylight time.
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0:12:32
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The only problem is, this says +120
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and this is actually in off set in minutes rather than hours.
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0:12:41
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+120, this is so that we can have half hours.
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0:12:47
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But +120, that two hours and that’s not proper,
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0:12:52
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that’s not what we were told to use.
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0:12:53
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We were told to use +1 hour.
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0:12:55
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So, we’re actually going to give it Western European standard time
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0:13:01
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or we could even give it GTB standard and daylight time.
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0:13:05
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We just have to make sure that it has a daylight time as well.
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0:13:10
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Like for instance, Eastern European, no, that has a daylight as well.
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0:13:14
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So, has to be one of these. So, we’ll just give type 23.
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0:13:18
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Time zone 23.
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0:13:22
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That’s not actually gonna download the name of the time zone.
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0:13:26
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So, that’s not going to be that important.
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0:13:28
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So, now, we need to do time format
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0:13:31
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and we’re told to set it to 24-hour format
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0:13:33
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and then date format and we were told to set it to month-day-year.
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0:13:41
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We’re told to set it to day-month-year, weren’t we?
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0:13:44
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Yes, day-month-year so we’ll say day-month-year.
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0:13:48
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We’re done. Do show run. Begin with voice service.
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0:14:00
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Here we got our voice service VOIP,
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0:14:02
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allow connection SIP to SIP turning on cue
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0:14:04
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but allowing phones to talk, SIP phones to talk to one another,
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0:14:08
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find all control for the source interface for SIP
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0:14:13
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and for media to loopback 0, turn on the register server
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0:14:18
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and actually go ahead and configure the bases of our voice register global.
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0:14:23
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Now, well, this is done, we’re going right this real quick.
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0:14:29
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If we do a show voice register, not global
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0:14:34
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but show voice register TFT bindings.
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0:14:37
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This is actually TFTP files that have been created by the SIP CME server
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0:14:43
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but there are none, why?
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0:14:46
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Well, one of the things we haven’t done is back in voice registrar global.
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0:14:53
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We didn’t say create profile.
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0:14:57
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This is why if you had ever done CME for Skinny,
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0:15:01
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that would be like create CNF files.
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0:15:04
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This is actually creating the configuration files.
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0:15:08
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So, now, if we do show voice register TFTP bind,
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0:15:12
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We've got some basic files that the server has generated in memory
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0:15:18
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where it’s running in the system.
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0:15:20
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We can look at that through not a show like we can do flash
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0:15:25
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So, we can’t do show system, it’s not an option.
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0:15:28
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But we can do DIR for directory flash or we can do DIR.
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0:15:34
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We got system, MB RAM and a number of other things.
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0:15:36
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We got DIR system
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0:15:40
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and we see that there are some configuration subfolders
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0:15:45
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D for directory. We've got CME and ITS CME.
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0:15:49
|
Let’s go ahead and just do that again.
|
|
0:15:51
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Let’s just do CME/.
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0:15:55
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We've got SIP phones so jump down another directory, SIP phones,
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0:16:01
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and we've got some SIP phone configuration files
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0:16:04
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that have been created dynamically in memory.
|
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0:16:07
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Also, DIR system/ITS
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0:16:13
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will show us the information for Skinny CME.
|
|
0:16:17
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So, it was originally named ITS, IP telephony system as I mentioned earlier.
|
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0:16:22
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So, it is still actually maintains that name for Skinny based files
|
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0:16:26
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and all SIP based files are under CME.
|
|
0:16:33
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So, again, let’s just do that show voice register TFTP bind
|
|
0:16:37
|
and notice it only has sink information.
|
|
0:16:40
|
SIP default, so it’s serving it as a root file
|
|
0:16:43
|
but the actual URL or the actual path is found here.
|
|
0:16:48
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This is the alias.
|
|
0:16:50
|
We've got a soft key default for KPML, key press mark up language .xml.
|
|
0:16:55
|
Key press mark up language is a mark up language from xml
|
|
0:17:00
|
and then the soft key default.
|
|
0:17:01
|
We don’t have any SIP config or even SEP MAC address config files
|
|
0:17:08
|
and that’s because we haven’t created any actual phones yet.
|
|
0:17:13
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So, that takes care of this task we’ve set up the basic CME SIP server.
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