The basics of IP telephony from Cisco or familiarity with Cisco CallManager Express
Most recently, a client came to our organization and asked to configure IP-ATS based on the Cisco 2921 platform and Cisco CallManager Express (CME) software. Customers scatter is not decent in our time, therefore, decided to provide this service.
I undertook this business not so much with the goal of cutting the dough for the company, but with the goal of getting to know how this Cisco IP-telephony is arranged, for the purpose of self-development, I mean.
Familiarized. The topic is nice; for me as for tsiskary, it is much more convenient to work, rather than with Asterisk.
')
You can write a lot about CME capabilities. Today I will tell you about the very basics.
Let's try to solve a simple problem: there are a couple of phones - one works on SIP, the other on SCCP, we need to make them call each other. How to do it - under the cut.
The zero point is the setting of the dhcp server on the router, since almost all phones by default are configured to receive an IP address in this way. If someone forgot or did not know, then in Tsiski it is done like this:
All settings seem to be obvious. Except, probably the 150th option. This option sends the address of the tftp server in the DHCP settings (I will not write about using the tftp server in this article, but the thing is very useful for working with phones)
By default, Cisco is very good with understanding SCCP protocol (due to the fact that this protocol belongs to Cisco itself), and is not very friendly with the more common SIP. So, for example, in order to simply allow calls between SIP phones connected to our PBX, you need to configure:
Next, configure the registration period for SIP phones:
Now you need to create a so-called. codec class - a set of codecs that you plan to use:
The first attempt will be made to use the 711alaw codec, if it fails, then g711ulaw, and then intuitively.
These settings will be enough.
All settings related to the operation of the SCCP protocol occur in the telephony-service section. Here is the minimum we need:
An ephone is such a thing, which, to put it simply, symbolizes the telephone, that is, the image of a telephone device in the concept of CME. The max-ephones parameter is responsible, respectively, for how many SCCP phones can be registered on this PBX.
dn - (directory number) is such a thing, which, to put it simply, symbolizes a telephone number. By analogy, it is clear what parameter max-dn is responsible for.
By the way, it is not very correct to specify too many ephone and dn , because CME will allocate random access memory for each of these “units” during initialization.
Plus, the number of supported ephone and dn depends on the platform model and software version.
ip source-address specifies the IP address (and therefore the interface) and the port on which requests from SCCP phones will be heard
All settings relating to the operation of the SIP protocol occur in the section voice register global . Here is the minimum we need:
Two differences. First, you need to execute the command mode cme , which seems to hint to the router that CME will now work with SIP phones (this is to the fact that not only the software described by us works with the SIP protocol from Cisco). Second, now instead of ephon, we will operate with voice-pools
You can start setting up the numbering and registration parameters of phones.
The technology in both protocols is the same and quite simple - dn is first created, then a “phone” is created, and its number (or several) is attached to the phone.
In our example, we will use their MAC addresses to authorize phone sets on the server. This is logical, since in practice the same telephone and telephone number are assigned to a specific person, and a person usually rarely moves his workplace in the office, and if he does, he carries the telephone with him.
Create a phone number:
We created the phone number 100, the description inside the CME is “Beauty-User”, the name displayed on the phone will be “The Real Man”
Create an image of the phone. we bind it to the MAC address of the real device, we bind the number to it:
The binding of the telephone set (or rather the telephone line of our apparatus) is performed by the button 1: 1 command. The format is as follows - first we type the key word “button” , then we indicate the number of this “button” (the first one in our example), then we indicate the action that needs to be done (“:” - a colon means that now we will tie some phone number) and then the parameter corresponding to the action - in our case we indicate the phone number (that is, the number of the dn pre-configured).
You probably saw a tsiskofon in life \ on the pictures? Seen there are all sorts of buttons near the screen? These are the very buttons. In fact, each of these buttons is responsible for its own line. And you can perform a lot of actions on them (not only to bind to any number). But it is worthy of a separate article.
In the meantime, we can safely turn on the phone with the specified MAC address to the network, wait for it to boot, see on its small screen near the first button the number 100 and the cherished name - “The Real Man”. If we pick up the phone, then we hear the coveted buzzer
The directory number is created as follows:
I think everything is clear. We describe further the parameters for the phone:
Binding to the first line of the first number is the command number 1 dn 1 .
Using the voice-class codec 1 command, we define a set of acceptable codecs for the phone (we described this set at the beginning).
Next, use the username \ password command to set the authentication data.
In order for the phone to register, you need to go to its web-muzzle and in the settings of the first line specify the server address - 192.168.101.1 and authentication data. We save.
Reboot.
...
PROFIT!
The IP address of the phone can be learned from the information from the dhcp-pool for Mac using the
It seems to be all. Were described the minimum CME settings so that it can register two phones (SIP and SCCP), having received the address via DHCP.
The main source of knowledge for a start was a video course to CCNA Voice from Jeremy Cioara. Quite a living language (English truth) tells about IP telephony in general and about the nuances of configuring this very CME. True, there is not a word about the setting of SIP phones.
A very good manual, of course, is on cisco.com. It is called the Cisco Unified Communications Manager Express System Administration Guide. Available here . There is also a link to download to PDF.
In the future, I will try to talk about the various nuances of setting up IP telephony on Cisco in general, and on CME in particular - setting up a tftp server for uploading firmware to phones, connecting lines from PSTN, specifics of CME installation and other features.
I undertook this business not so much with the goal of cutting the dough for the company, but with the goal of getting to know how this Cisco IP-telephony is arranged, for the purpose of self-development, I mean.
Familiarized. The topic is nice; for me as for tsiskary, it is much more convenient to work, rather than with Asterisk.
')
You can write a lot about CME capabilities. Today I will tell you about the very basics.
Let's try to solve a simple problem: there are a couple of phones - one works on SIP, the other on SCCP, we need to make them call each other. How to do it - under the cut.
Initial CME Setup
The zero point is the setting of the dhcp server on the router, since almost all phones by default are configured to receive an IP address in this way. If someone forgot or did not know, then in Tsiski it is done like this:
CME#conf t
CME#(config)ip dhcp pool voicepool
CME#(config-ip-dhcp)network 192.168.101.0 255.255.255.0
CME#(config-ip-dhcp)default-router 192.168.101.1
CME#(config-ip-dhcp)option 150 ip 192.168.101.1
CME#(config-ip-dhcp)lease 0 0 30
All settings seem to be obvious. Except, probably the 150th option. This option sends the address of the tftp server in the DHCP settings (I will not write about using the tftp server in this article, but the thing is very useful for working with phones)
By default, Cisco is very good with understanding SCCP protocol (due to the fact that this protocol belongs to Cisco itself), and is not very friendly with the more common SIP. So, for example, in order to simply allow calls between SIP phones connected to our PBX, you need to configure:
CME#conf t
CME#(config)voice service voip
CME#(config-voice)allow-connections sip to sip
Next, configure the registration period for SIP phones:
CME#conf t
CME#(config)voice service voip
CME#(config-voice)sip
CME#(config-voice-sip)registrar server expires max 3600 min 3600
Now you need to create a so-called. codec class - a set of codecs that you plan to use:
CME#conf t
CME#(config)voice class codec 1
CME#(config-voice)codec preference 1 g711alaw
CME#(config-voice)codec preference 2 g711ulaw
CME#(config-voice)codec preference 3 g729br8
The first attempt will be made to use the 711alaw codec, if it fails, then g711ulaw, and then intuitively.
These settings will be enough.
Setup for SCCP operation
All settings related to the operation of the SCCP protocol occur in the telephony-service section. Here is the minimum we need:
CME#conf t
CME#(config)telephony-service
CME#(config-telephony)max-ephones 10
CME#(config-telephony)max-dn 10
CME#(config-telephony)ip source-address 192.168.101.1 port 2000
An ephone is such a thing, which, to put it simply, symbolizes the telephone, that is, the image of a telephone device in the concept of CME. The max-ephones parameter is responsible, respectively, for how many SCCP phones can be registered on this PBX.
dn - (directory number) is such a thing, which, to put it simply, symbolizes a telephone number. By analogy, it is clear what parameter max-dn is responsible for.
By the way, it is not very correct to specify too many ephone and dn , because CME will allocate random access memory for each of these “units” during initialization.
Plus, the number of supported ephone and dn depends on the platform model and software version.
ip source-address specifies the IP address (and therefore the interface) and the port on which requests from SCCP phones will be heard
Configure CME to work with SIP phones
All settings relating to the operation of the SIP protocol occur in the section voice register global . Here is the minimum we need:
CME#conf t
CME#(config)voice register global
CME#(config-voice)mode cme
CME#(config-voice)source-address 192.168.101.1 port 5060
CME#(config-voice)max-dn 50
CME#(config-voice)max-pool 36
Two differences. First, you need to execute the command mode cme , which seems to hint to the router that CME will now work with SIP phones (this is to the fact that not only the software described by us works with the SIP protocol from Cisco). Second, now instead of ephon, we will operate with voice-pools
Registering phones
You can start setting up the numbering and registration parameters of phones.
The technology in both protocols is the same and quite simple - dn is first created, then a “phone” is created, and its number (or several) is attached to the phone.
In our example, we will use their MAC addresses to authorize phone sets on the server. This is logical, since in practice the same telephone and telephone number are assigned to a specific person, and a person usually rarely moves his workplace in the office, and if he does, he carries the telephone with him.
Registering a sccp phone
Create a phone number:
CME#conf t
CME#(config)ephone-dn 1
CME#(config-ephone-dn)number 100
CME#(config-ephone-dn)description Beauty-User
CME#(config-ephone-dn)name The Real Man
We created the phone number 100, the description inside the CME is “Beauty-User”, the name displayed on the phone will be “The Real Man”
Create an image of the phone. we bind it to the MAC address of the real device, we bind the number to it:
CME#conf t
CME#(config)ephone 1
CME#(config-ephone)mac-address B4A4.E328.BDEC
CME#(config-ephone)button 1:1
The binding of the telephone set (or rather the telephone line of our apparatus) is performed by the button 1: 1 command. The format is as follows - first we type the key word “button” , then we indicate the number of this “button” (the first one in our example), then we indicate the action that needs to be done (“:” - a colon means that now we will tie some phone number) and then the parameter corresponding to the action - in our case we indicate the phone number (that is, the number of the dn pre-configured).
You probably saw a tsiskofon in life \ on the pictures? Seen there are all sorts of buttons near the screen? These are the very buttons. In fact, each of these buttons is responsible for its own line. And you can perform a lot of actions on them (not only to bind to any number). But it is worthy of a separate article.
In the meantime, we can safely turn on the phone with the specified MAC address to the network, wait for it to boot, see on its small screen near the first button the number 100 and the cherished name - “The Real Man”. If we pick up the phone, then we hear the coveted buzzer
We register SIP-phone
The directory number is created as follows:
CME#conf t
CME#(config)voice register dn 1
CME#(config-voice-register-dn)number 200
I think everything is clear. We describe further the parameters for the phone:
CME#conf t
CME#(config)voice register pool 1
CME#(config-voice-register-pool)id mac 1CDF.0F4A.152E
CME#(config-voice-register-pool)number 1 dn 1
CME#(config-voice-register-pool)voice-class codec 1
CME#(config-voice-register-pool)username test password test
Binding to the first line of the first number is the command number 1 dn 1 .
Using the voice-class codec 1 command, we define a set of acceptable codecs for the phone (we described this set at the beginning).
Next, use the username \ password command to set the authentication data.
In order for the phone to register, you need to go to its web-muzzle and in the settings of the first line specify the server address - 192.168.101.1 and authentication data. We save.
Reboot.
...
PROFIT!
The IP address of the phone can be learned from the information from the dhcp-pool for Mac using the
show ip dhcp binding command, either in the settings of the phone itself - voice-activated, or visually the menu.It seems to be all. Were described the minimum CME settings so that it can register two phones (SIP and SCCP), having received the address via DHCP.
Sources of knowledge
The main source of knowledge for a start was a video course to CCNA Voice from Jeremy Cioara. Quite a living language (English truth) tells about IP telephony in general and about the nuances of configuring this very CME. True, there is not a word about the setting of SIP phones.
A very good manual, of course, is on cisco.com. It is called the Cisco Unified Communications Manager Express System Administration Guide. Available here . There is also a link to download to PDF.
In the future, I will try to talk about the various nuances of setting up IP telephony on Cisco in general, and on CME in particular - setting up a tftp server for uploading firmware to phones, connecting lines from PSTN, specifics of CME installation and other features.
Source: https://habr.com/ru/post/124411/
All Articles