Basic asterisk setup after installation. Asterisk: installation and instructions

The Linux program Asterisk is needed to create telephony and PBX servers. She has a lot useful functions, which will be appropriate in any office. Conferences, answering machines, call distribution, text messages, video, voice menu and voice mail. The utility supports popular communication protocols. They can be switched. But installing Asterisk on Ubuntu is not the easiest thing. The application is downloaded and configured via the terminal. Therefore, you need to learn console commands.

Installed Asterisk must be configured both through the menu and through the terminal. So open your console.

For now, to work with Asterisk you need superuser rights. But it's better to create individual user and give him access to the program's functions. To do this:

The utility is ready to use. Next steps depend on the conditions under which the station will be used. There are several files that contain the configuration:

• conf - general settings server. In it you can indicate how many subscribers are connected. They establish dialing rules.
• conf - it also configures dialing rules and sets call parameters.
• conf - configuration of subscriber accounts.

With Asterisk you can create a PBX of any complexity. It has many useful functions and modules. After installation and configuration, you will be able to interact with the station using the client software. For example, Linphone.

Websites, forums and entire portals like voip-info.org are dedicated to setting up Asterisk. But there are always people for whom this topic is a dense forest, and the product needs to be launched “yesterday”. Just a month and a half ago, all I knew about Asterisk was that it existed. This work is intended only to make life a little easier for those who find themselves in the same situation after me. It may contain errors of any degree. If you are an Asterisk configuration guru, point out the error, I will correct it, and my descendants will be grateful to you.

At some point in time, through my efforts, the company’s infrastructure development strategy was literally turned upside down. Goals and objectives were outlined and it became clear that we need a dynamic, mobile server infrastructure in order to save money, so that in case of any problems we can quickly recover and/or rise in another place, so that it is easier to care for the hardware, etc. p.

In general, on paper everything was beautiful and easy. It was just a matter of deploying and configuring Asterisk.
First, two simple servers were purchased. Core i7-930, 12GB of memory and 12TB of SATA disks. Free Xen Server was installed on these servers and deployed virtual servers and workstations for remote workers.
A QNAP NAS, which was inherited, is used as external storage.
Terabyte local storage needed only to store the media bank, which is quite large. The servers themselves collectively weigh less than 3 terabytes.
One of the servers was Asterisk to replace the old Panasonic TDE-200. Because the company needs cheap communication, direct numbers to different countries, and free negotiations between employees, then in general there is no alternative to it for 12 shekels.

It was decided to configure Asterisk natively for Citrix Xen operating system CentOS. The same one, only from the side. I’ll make a reservation right away: I’ve been a Windows administrator all my life, and I’ve only “played” with *nix in virtual machines, not counting a couple of web servers on hardware. Therefore, this deployment of Asterisk in Citrix Xen smacks of some naivety from the point of view of experienced people.
In general, I followed the proven path: a detailed how-to instruction for dummies from the official manuals is opened on the laptop and executed on the server literally and without initiative. It will be possible to play as hero-fixers later, when everything is configured and working in the “default” state and the first full backup is made.

Not that I was very surprised, but, nevertheless, I went to a telecommunications friend for advice on setting up Asterisk when I discovered that there were already three living Asterisk forks: 1.4, 1.6 and 1.8. Each of them has their own cockroaches and lotions. Because I wasn’t flattered by the hope of defeating asterisk on a whim, so I decided to install version 1.4, like a friend’s, in order to have the highest possible compatibility of configs. Although, subsequently, we had to upgrade to 1.6 due to the lack of TCP support in 1.4, when such a need arose.
To do this, a digium repository was attached to CentOS and Asterisk was installed from it.
The installation went as expected, without a hitch. Basically, a sensible step-by-step roadmap with voip-info was used.
iptables was immediately corrected to allow traffic on SIP ports.
Not immediately, but I recommend that future generations install and configure a package called fail2ban immediately. Not at all extra program on a server exposed to the Internet. A day after installation, all of Taiwan went into drop, judging by iptables.

Let's start with the fun part - planning. Before any configuration process, it is necessary to write down and structure all the customer’s wishes and be sure to sign an agreement in order to avoid future conflicts of the type “no, I said something completely different.” In my case, the customer is the company I work for, but this does not change anything. We write terms of reference on the functionality, discuss it with the responsible people, settle disagreements and sign the final version and begin setting up Asterisk.
The specifications for the configuration turned out like this:

• Phone numbering is four digit.
• The first digit of the number is a sign of the country where the office is located.
• Numbering in each department begins with the first digit of the next hundred.
• Each department is a separate call group.
• If the called subscriber is in any way unavailable for the call, the call is transferred to the entire group.
• Logic for servicing incoming calls:
• Pick up the phone.
• Play greeting.
• Play an advertisement about current promotions.
• Play options for calling various departments or offer to dial the subscriber’s number.
• Starting from point 2, wait for any commands to be entered.
• After completing step 4, wait another 10 seconds for commands to be entered.
• If there are no commands, play the call options again.
• If the subscriber is still dumb, then transfer the call to the secretary.
• For users who need such a service, screw voicemail. In this case, if the user is in any way unavailable for calling, transfer the call to voicemail.
• By default, calls are allowed only to Russia and Italy - the countries where our company mainly operates. The remaining directions are opened as needed with the narrowest masks, because we all remember scary stories about “Cuban telephone operators” and we don’t want one day our PBX suddenly starts calling Sierra Leone on all 50 channels at $14 per minute for each channel.

At first glance, the task is simple and setting up Asterisk will not take much time, but in practice it took quite a lot of blood to coordinate it. Especially, to agree on a ban on all directions except those that are clearly permitted. The client had to agree to this only in the face of the otherwise inevitable need to configure a VPN on every device connecting to the PBX. I just try to keep the configurations as simple and understandable as possible, and the client always thinks that adding some functionality to the system is “just a couple of buttons to press.”

Now comes the fun part: the process of setting up Asterisk.
In order not to drag out an already long article, burdening it detailed description torment of the initial configuration and subsequent tuning, I’ll just show my current configs with comments about what’s what.

context=default ;default context
allowguest=no ;no guests, we are not expecting anyone
allowoverlap=no ;This parameter determines whether or not we allow the transfer of a call with additional digits. We do not allow it.
alwaysauthreject=yes ;This parameter means that all errors for the external client look the same - authorization error. Designed to complicate hacking of PBX.
useragent=Orgue de Barbaris ;This is how we introduce ourselves to others. I identified myself by the name of the first gramophone to make it easier for technical support of providers to find dialogues with my PBX, because... I have to communicate with them periodically.
defaultexpiry=360 ;Interval for mandatory re-registration of clients.
callevents=yes ;This is for the future so that the client can play music on hold controls
limitonpeer=yes ;We count outgoing and incoming calls from peers together.
tcpenable=yes ;Allow TCP protocol for SIP.
rtptimeout=60 ;How long a client may not respond before being kicked out of the system.
language=ru ;System language. Used for voice menu.
bindport=5060 ;Default port. It can be changed to complicate the calculation and hacking of the PBX, but the solution is controversial.
bindaddr=0.0.0.0 ;I have zeros because... The PBX has two interfaces and both have clients.
srvlookup=yes
tos_sip=cs3 ; http://www.voip-info.org/wiki/view/Asterisk+sip+tos
tos_audio=efdisallow=all ;first disable all codecs
allow=alaw ;then, we allow only those that we want to use. The order doesn't matter.
allow=g729
allow=g723
allow=ulaw

canreinvite=no ;This parameter prevents clients from communicating directly. Communication occurs only through the server. I do it this way because mobile clients use paid clients with the g.729a protocol, and landline clients use free clients without it. Therefore, I preferred to strain the server by transcoding alawg.729a on the fly in favor of connection quality for mobile clients. Moreover, the Core i7-930 does not notice this load at all.

This article is a note in case I need to remember how to configure Asterisk to receive and send calls. This note should be enough to configure an office station on ip pbx Asterisk. Until now, I did not have a strong idea or experience of how to configure Asterisk to work without using FreePbx, that is, using only configuration files and the console. And now I’m just learning this. On the Internet, all settings come down to solving one problem or another, but I did not see an article capable of providing at least a basis for minimal configuration of an IP station. Maybe I didn't search well.

So, I built and installed asterisk in the /etc/asterisk folder configuration files. What's next?

1. The ability for internal subscribers to make calls to each other

2. Setting up external lines in order for the station to receive the service of accessing city and long-distance numbers and receiving incoming calls

3. Configuring the ability to make calls to subscribers to city and long-distance numbers. And take calls from the city.

Here are three main tasks that need to be solved; naturally, the scheme may be different. That is, subscribers can be united into groups, there can be several external telecom operators, incoming call can be sent to an autoinformer, a queue, or maybe both. The scheme of services may vary and become more complex, but the first three conditions will be met in any case.

Therefore, the first thing we do is create internal subscribers (peers) in the sip.conf file

Subscriber A will have the following settings:

Disallow=all
allow=ulaw
allow=alaw
type=friend
username=101
secret=password
qualify=yes
context=101
callerid=101
host=dynamic
call-limit=1
canreinvite=yes
accountcode=101
amaflags=billing
deny=0.0.0.0/0
permit=192.168.0.66

Where:
disallow=all , prohibit the use of all codecs except G.711
allow=ulaw
allow=alaw

type=friend, this parameter can take three values:

- peer, this type is used to make incoming and outgoing calls. For incoming calls, the username and password will never be checked, but only the IP address and port number of the call source.

- user an entry that can only make incoming calls

- friend an entry that is both peer and user
username=101
secret=password
Login and password respectively

Periodically send a SIP message of the Options type to check monitoring of device availability for registration, useful when working through NAT to save values ​​in the translation table

Context name, with routing rules

Caller ID of our number

host=dynamic
host ip from which registration will take place

call-limit=1
(callcounter)
Limit simultaneous calls, here you can set the number of channels

canreinvite=yes
(directmedia)

This parameter is responsible for the RE-Invite procedure, if no, then all calls will go through Asterisk, if yes, the RTP stream will be sent between two devices bypassing Asterisk

accountcode=101
amaflags=billing

The CDR record for the accountcode field will have the value 101
The value of the AMA flag field for outgoing calls from this SIP user

deny=0.0.0.0/0
permit=192.168.0.66
ACL ip, who we will allow registration

Similarly for subscriber B:

Disallow=all
allow=ulaw
allow=alaw
type=friend
username=102
secret=password
qualify=yes
context=102
callerid=102
host=dynamic
call-limit=1
canreinvite=yes
accountcode=102
amaflags=billing
deny=0.0.0.0/0
permit=192.168.0.67

Some more options in the feast that may be useful:

transport=tcp,udp
I can work not only using udp, it requires setting tcpenable=yes in general
rtptimeout=180
rtpholdtimeout=420

Useful for SIP clients that do not Hangup when closed or disconnected. IN in this case The RTP session will be able to reset itself by timeout, rtpholdtimeout - if the call is on hold

nat=force_rport,comedia

If the device is behind NAT, then we fix the work via rport, send RTP packets back to the IP address and port from which they were received, ignoring information from SDP

defaultexpiry=60
Registration expiration time in seconds for incoming and outgoing registrations

dtmfmode=rfc2833

How we will work with DTMF signals

1. So, task one, setting up routing between sip clients 101 and 102, in the simplest version can be solved with one line in the sip.conf file and include configs

exten => _XXX,1,Dial(SIP/$(EXTEN),20,T)

It is convenient to use sip.conf like this:

Include => in_out_peer_local

Now subscriber 101 will be able to communicate with subscriber 102 and vice versa, 102 will be able to call 101

2. To create conditions under which internal subscribers will be able to access external lines, we will connect the station to the telecom operator using the same sip

That is, we will create a trunk
type=peer
host=XX.XXX.XXX.XX
port=5060
fromdomain=ip of our station
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
;allow=g722
trustpid=yes
sendrpid=yes
dtmfmode=rfc2833
qualify=yes
insecure=invite,port
deny=0.0.0.0/0.0.0.0
permit=XX.XXX.XXX.XX/255.255.255.255
context=provide

The trunk settings are very similar to the peer settings, in principle this is a peer, important options
fromdomain – set fromdomain in the sip packet so that you can understand where to return sip packets to our station
trustpid=yes
sendrpid=yes

To trust or not the received Remote-Party-ID

insecure=invite,port

We do not require authentication of incoming INVITE messages, we do not check the port number only for compliance with the IP address

3. It’s just a matter of small things, as soon as the trunk is activated, you need to register outgoing routing to make calls to subscribers A and B
The de facto standard seems to say to always set up outgoing communication first

Creating a context for making outgoing calls

For outgoing calls to the city

exten => _XXXXXX,1,Dial(SIP/ provider /$(EXTEN))
exten => _XXXXXX,n,Hangup

For outgoing calls to long-distance destinations

exten => _8.,1,Dial(SIP/ provider /$(EXTEN))
exten => _8.,n,Hangup

After the outgoing connection is working, we set up the incoming connection

In a simple case without autoinformer, groups and queues it will look like this

exten => 2466754,1,Dial(SIP/$(EXTEN))
exten => 2466754,n,Hangup

Any station can be configured in the same way according to the principles discussed above.