Installing and configuring Asterisk telephony in Linux Ubuntu. IP telephony from scratch. Installing AsteriskNOW

Asterisk is the solution for computer telephony open source that supports VoIP, SIP and PJSIP protocols for the Linux operating system. If you connect the appropriate equipment to your computer, then using Asterisk you can create full-fledged station ATC. Among the main features of the program are voice mail, conferences, interactive voice menu and call distribution.

In this article we will look at how to install Asterisk 13 on Ubuntu 16.04. The newest version of the program is not in the official repositories, since it is not used by ordinary users so often, so we will have to build it from source.

To build from sources, we will need to perform most of the actions on behalf of the superuser, so we will immediately get full rights in the system:

Please note that if you log into your server via ssh as root, you will still have to connect as regular user, and only then switch to the root console. Next you need to install dependencies for the program to work:

apt-get install build-essential wget libssl-dev libncurses5-dev libnewt-dev libxml2-dev linux-headers-$(uname -r) libsqlite3-dev uuid-dev git subversion

cd /usr/src
$ wget downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz

To unpack the sources, type:

tar zxvf asterisk-13-current.tar.gz

Before compiling the program, we need to build and install pjproject. First, download the sources:

git clone git://github.com/asterisk/pjproject pjproject
$ cd pjproject
$ ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS="-O2 -DNDEBUG"
$ make dep make && make install
$ldconfig
$ ldconfig -p|grep pj

Finally, we are now ready to build and install Asterisk on Ubuntu 16.04:

cd .. contrib/scripts/get_mp3_source.sh
$ contrib/scripts/install_prereq install

These commands will allow you to install some dependencies for playing mp3. It may take some time to complete and you will also need to enter a country code. Next we begin the assembly:

./configure && make menuselect && make && make install

When the installation is complete, run the following command to avoid creating hundreds of configuration files manually:

To install init scripts that will allow asterisk to start automatically, run:

make config ldconfig

Now you can start the asterisk server and make sure that everything really works:

service asterisk start

And look at the information about the utility through the terminal:

If everything was done correctly and the installation of Asterisk on Ubuntu 16.04 was successful, then everything will work.

The program runs as superuser. But we need it to be able to run as the asterisk user, so we need to do a few more settings. This way your system will be more secure. First you need to stop the running service:

groupadd asterisk
$ useradd -d /var/lib/asterisk -g asterisk asterisk

Next we need to change the initialization script so that the program is launched as the desired user. We can edit the /etc/default/asterisk file manually, but it is easier to use the following commands:

sed -i "s/#AST_USER="asterisk"/AST_USER="asterisk"/g" /etc/default/asterisk
$ sed -i "s/#AST_GROUP="asterisk"/AST_GROUP="asterisk"/g" /etc/default/asterisk

In order for everything to work correctly, the asterisk user needs to be given access to all program directories:

chown -R asterisk:asterisk /var/spool/asterisk /var/run/asterisk /etc/asterisk /var/(lib,log,spool)/asterisk /usr/lib/asterisk

We will also indicate correct settings user for the runuser and rungroup fields in the asterisk.conf file:

sed -i "s/;runuser = asterisk/runuser = asterisk/g" /etc/asterisk/asterisk.conf

$ sed -i "s/;rungroup = asterisk/rungroup = asterisk/g" /etc/asterisk/asterisk.conf

When everything is ready, start the previously stopped service or reboot if you haven’t stopped it:

sudo service asterisk start

Then we check:

If the user and group are asterisk, then we did everything right.

That's all. Installation of Asterisk 13 on Ubuntu 16.04 is complete. Now you can proceed to further network setup to make calls between your subscribers directly via the Internet. As client software, you can use any program that supports the required protocols, for example Jitsi or Linphone.

Founder and administrator of the site, I am passionate about open software and the Linux operating system. I currently use Ubuntu as my main OS. In addition to Linux, I am interested in everything related to information technology and modern science.

Good afternoon, in this article I will tell you how to install the most popular server VoIP telephony Asterisk. By using of this server You can organize telephony in the office, at home, or implement automatic notification, but I will talk about this in other articles. And so let's start installing asterisk on the ubuntu server.

Installing Asterisk on Ubuntu Server is extremely simple, you only need one command:

The first thing after installing Asterisk is to disable the ability to receive calls anonymously. To do this, open the sip.conf file and uncomment the line allowguest=no, since by default it is Yes. By the way, to search for a given line in nano, you can press Ctrl+W and enter the desired part of the line and press Enter; if there are several such words in the text, then to continue the search, press Ctrl+W and Enter.

In the same file we can indicate VoIP providers and our internal subscribers. I use the operator it-sekret, since it has cheap calls it-sekret.ru. To do this, add at the end of the file

For the operator:

Type=peer context=it-sekret host=xxx.it-sekret.ru username=USERNAME secret=PASSWORD fromuser=USERNAME nat=force_rport insecure=port,invite

If the operator needs constant registration, for example for incoming calls, then you need to write this line

Register => userID: [email protected]/userID

Where userID is the username issued by the operator and PASSWORD is the password, xxx.it-sekret.ru the server address issued by the operator.

Create an internal subscriber:

Type=friend context=internal host=dynamic username=1001 secret=1234

After adding these entries, save the file and exit. In order to apply the settings, you must either restart the server or tell it to re-read the settings. This method is better, because if your server is active, then the calls that are currently in progress will not be interrupted.

Now let's set up the phone or softphone. I'm using X-lite softphones quite a lot, you can use it to your taste. Configure X-lite settings as shown in the screenshot.

To check the correctness of the settings in the Asterisk CLI (hereinafter simply CLI), run the sip show peers command

First of all, let's make a copy of the original extension.conf file

Now in extension.conf we will write a rule for an outgoing call.

In the CLI, run the reload command to apply the settings. And now let's try to make a call.

The call went through, now we need to set it up incoming call. To do this, you need to write a dialing rule.

On this easy setup ends, in the following articles I will tell you how to record conversations, set up IVR, make a CallBack from the site and much more.

23.02.2012 15:56

An article that allows novice programmers to perform basic asterisk setup

Asterisk is one of the most common IP telephony servers, along with systems such as Switchwox and 3CX. This system has gained great popularity with the spread of high-quality Internet in our country, as well as many telephone providers providing telephone numbers in SIP format or so-called SIP numbers. One of such companies is Datagroup, which has established itself as one of the leaders in the communication services market. With SIP numbers becoming cheaper, it makes sense to use modern telephone systems IP telephony instead of the old analogue mini-PBX. This is due to the wide functionality call centers based on Asterisk. At the same time, the cost of equipment and setting up Asterisk is practically not inferior in monetary costs.

Installing asterisk is done by downloading and then compiling the freely distributed source code developed by Digium. asterisk sources exist only for unix-compatible operating systems, the most common of which are Linux and FreeBSD. Installing asterisk under Windows is only possible by implementing a virtual machine. You can download the latest versions of asterisk from the following link. At the moment, 4 versions of this server are being developed. The versions asterisk 1.4 and asterisk 1.6 have the greatest stability of all modules. On the other hand, in terms of functionality, version asterisk 1.8 compares favorably, which has support for many new modules and functions, including support for the SRTP protocol, work with IPv6 and improved performance when working with SIP and ISDN.

This article will discuss installing asterisk v 1.6.22 on Linux Fedora Core.

To start the installation, download the asterisk sources. To do this, run the commands in the server console:

# wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.22.tar.gz

# tar -xzvf asterisk-1.6.2.22.tar.gz

Before compiling the sources, let’s install all the necessary accompanying modules and utilities:

# yum install gcc

# yum install gcc-c++

# yum install make

# yum install libxml2-devel

# yum install ncurses-devel

After all additional utilities are installed, we install asterisk itself:

# cd asterisk-1.6.2.22

After successful installation, in the /etc/asterisk directory you will find the asterisk"a configuration files. Asterisk is configured by editing configuration files that have a very specific structure and operators. The basis for the configuration when configuring asterisk is the sip.conf and extensions.conf files. Having studied and edited these files can be achieved proper operation server.

The sip.conf file includes the settings of all SIP devices registered on the server, which subsequently interact with each other by connecting two or more SIP channels, thereby initiating a telephone connection - that is, a call.

Example sip.conf file:

qualify=yes
nat=no

defaultexpiry=3600
disallow=all
allow=alaw
allow=ulaw
fromuser=044XXXXXXX
host=80.91.169.2
insecure=invite,port
secret=xxxxxxxxxxxx
type=peer
username=044XXXXXXX
context=incoming
language=en
qualify=yes
dtmfmode=auto

type=friend
host=dynamic
insecure=invite,port
username=fxo
secret=xxxxxxxx
context=incoming
qualify=yes
nat=no
canreinvite=no
allowtransfer=no
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw

type=friend
regexten=101
secret=xxxxxxxxxxxxx
context=internal
host=dynamic
callerid="101"
disallow=all
allow=alaw
allow=ulaw
language=en
callgroup=1
pickupgroup=1
qualify=yes
canreinvite=yes
call-limit=4
nat=no

This example shows simple list sip devices. It is easy to notice that the configuration contains blocks enclosed in square brackets. It is these blocks that describe the configuration of any devices (peers) when setting up Asterisk. Let's briefly go through the blocks and the most basic parameters.

The block is required. It includes settings and options that apply to all blocks by default. In other words, if a parameter is not specified when describing a device, then Asterisk reads it from this block.

The next three blocks directly describe SIP devices. For clarity, all three devices have different purposes. The block describes the parameters for the sip number of the Datagroup company, the block describes the parameters for connecting to a VOIP gateway, and the block describes the parameters of a regular internal sip device that can be used by a software or hardware IP phone.

Let's go over the main parameters of SIP devices. The context parameter determines which action algorithm to apply to a particular device. Callerid defines the peer number and name, which will be displayed when an outgoing call is made from it. The disallow and allow options determine which codecs to use for communication. The secret option is the password for registering this peer. It is advisable to make passwords sufficiently complex to avoid hacking by ill-wishers.

The extensions.conf file describes the interactions between contexts and communication channels, of which there can be a large number. In fact, this file is the main one when configuring asterisk.

Example extensions.conf file:

exten => _XXX,1,Dial(SIP/$(EXTEN),120,tT)
exten => _XXX,n,Hangup

Exten => _X.,1,Dial(SIP/datagroup/$(EXTEN),120,tT)
exten => _X.,n,Hangup

exten => _X.,1,Dial(SIP/101,60,tT)
exten => _X.,2,Hangup

Exten => h,1,Hangup

In this simplest example, we have two algorithms of actions (context) - internal and incoming. Let us recall that binding to contexts is carried out when creating peers in the sip.conf file and is subsequently assigned in extensions.conf when attempting to initiate communication with a peer.

Let us describe the two contexts in more detail. Just by their names it becomes clear that they are used for algorithms for internal subscribers and incoming calls. A generalized example, which can be expressed in the following line, will help you understand the design of this file:

Exten => _XXX,n,Command(Options)

These options are deciphered as follows:

Exten => _XXX is a service prefix with which almost all lines in this file begin. It describes the representation of the incoming context. In other words, if this context was contacted by peer number 101, then the algorithm for it should begin with the construction exten => 101. The prefix supports regular expressions and can be specified in a general way. For example, exten => _XXX indicates that under this algorithm all peers that consist of three digits are included.

N - line number of the algorithm. In any program code there is a sequence of statements that are executed in turn. In the same way, there is such a sequence in the asterisk settings. The only difference is that the commands executed by asterisk are numbered. It is mandatory to indicate the first line explicitly. As you may have noticed, all contexts begin with the lines:

Exten => _XXX,1,…..
exten => _X.,1,…..
exten => h,1,……

But further lines can be numbered implicitly, that is, like this:

Exten => _XXX,n,…..
exten => _X.,n,…..
exten => h,n,……

When the line number is implicitly specified, asterisk executes them sequentially from top to bottom, as it does with any other code.

Command - a command executed by asterisk. There aren't that many basic commands. For basic setup, it is enough to know the syntax of the Dial command, using which we will look at the remaining options as an example.

For example:
exten => _X.,1,Dial(SIP/101,60,tT) - asterisk will dial extension 101 and try to reach the subscriber within 60 seconds with the possibility of intercepting the call from the calling and receiving peers (tT options)
exten => _X.,1,Dial(SIP/datagroup/$(EXTEN),120,tT) asterisk will dial the number that is passed in the $(EXTEN) variable using the datagroup peer previously specified in sip.conf

The first example was used in the incoming context. This means that all incoming calls coming to this context are redirected to the internal subscriber who answers these calls. The second example, in turn, shows how outgoing call passing the dialed number in the service variable $(EXTEN).

Example files can be directly copied from this article and pasted into your sip.conf and extensions.conf and edited some options (logins, peer names, passwords, etc.). After setting up asterisk, we start the server with a simple command:

Further actions can be performed from the asterisk command line, which is launched with the command:

# asterisk -rvvvvd

Once in the console, you can check the operation of your server. It will display all the actions it performs.

P.S. If you were unable to configure the server, or you need a more complex configuration, we will be happy to help you configure asterisk. Our contacts.

Each SIP client or server is defined in this file as a text block, similar to the one below:

type=yyy
parameter1=value
parameter2=value

Where xxx is the name associated with the SIP client, or it can be an arbitrary SIP device name that can be referenced from other configuration files. Typically, if a SIP phone has an extension number of 123, then the corresponding settings in this file begin with the line . Please note that in order to be able to call this extension 123, you must describe it in your dial plan. could be: "user", "peer" or "friend".

The Asterisk server determines whether an incoming call matches a device name, for clients with type=user, based on the From: username header (ignoring the SIP domain). Another way to process incoming SIP requests is to determine if they match a section in this file, using the IP address of the incoming request to determine who the request came from, and looking for the appropriate Host= parameter value in the user section. If this parameter is set to Host=dynamic, then there will be no matching matches until the SIP client registers with the server.

Format:
register => user [:secret[:authuser]] @host [:port]

Example:
; Register 2345 with the sip provider, like number 1234 on our side.
register => 2345: [email protected]/1234

user - user ID used for the SIP server (for example, 2345) authuser - optional user name for authorization on the SIP server secret - user password host - domain or host name of the SIP server. This SIP server must be defined in its own section of the sip.conf file, where its parameters must be specified (mysipprovider.com). port - to which port number to send requests for registration on the host server. By default - 5060 /1234 - extension number for receiving calls in your Asterisk. 1234 - inserted into the SIP header contact, SIP registration request. This extension is used by a remote SIP server when it needs to make a call to your Asterisk. See the examples below. By default, the contextual "s" is used.
This, of course, is all well and good, but using unencrypted passwords in a text file is not a good idea, but what else can you do now?
You only need to register if: a) it must be possible to call you, and b) one of the parties has a dynamic IP address. You can check whether your server has successfully registered using the CLI command: "SIP SHOW REGISTRY", similarly, you can get a list of clients registered on your server using the command: "SIP SHOW PEERS". You can view more detailed information about a registered client using the command: "SIP SHOW PEER". Run the "HELP SIP" command in the CLI console to get a list of additional commands.

The server definition for making outgoing calls should be something like this:

type=peer
secret=password
username=2345
host=sipserver.mysipprovider.com
fromuser=2345
fromdomain=fwd.pulver.com
nat=yes
context=from-mysipprovider ; this contest must be defined in extensions.conf

In the extensions.conf file, to make outgoing calls, you must have a dialing rule that looks something like this:

Exten => _9.,1,Dial(SIP/$(EXTEN:1)@mysipprovider-out,30,r)

Please note that the $(EXTEN:1) construct retrieves the entire contents of the variable that contains the extension being called (matching the pattern), except for the first digit, in this case: 9 + a set of digits. Refer to the section on working with substrings in the description for more detailed information.

exten => 1234,1,Answer; 1234 - extension from contact information, default - "s"
exten => 1234,2,Dial(SIP/111,25,Ttr) ; incoming call redirect to SIP phone number 111
exten => 1234.3,Hangup

= : Allowed codecs, the order in which the codec is selected is determined by the order in which they are described in this command (First use: DISALLOW=ALL, before allowing any specific codecs) allowtransfer= yes | no: Appeared starting from version 1.4.0. When set to "no" - prohibits all types of call transfers. (except for those permitted in the user settings description). =all:Disables the use of all codecs (global setting) allowguest = yes (default) | no: Allow or disable guest calls (default - yes, you can specify the "osp" parameter as the value if asterisk is built with OSP support) = yes | no: If enabled, anyone can use the server as a peer (without checking for access; can be convenient when working with SER SIP proxy). bindaddr = 0.0.0.0:IP address to which Asterisk will receive IP packets of SIP calls bindport=5060:port number to which Asterisk will receive IP packets of SIP calls callerid = : Information for Caller*Id, used when there is none of those , which could be used as this value. The default is "asterisk". (The ability to override the default value is available in Asterisk 1.0.9. I'm not entirely sure about other versions.) = update | yes | no (global setting). For some reason, the default value is "Yes", be careful... context = : This is the default, which is used when the client does not have its own defined. This client-specific is used to route calls from that client to the desired destination. The content is described in the dial plan file - extensions.conf. defaultexpirey=120: Duration of the incoming or outgoing registration period. : inband | info | rfc2833 (global setting) = domains: Comma separated list of domains that the Asterisk server is responsible for. (appeared in Asterisk 1.2.x) = 200.201.202.203: IP address that will be used in SIP messages if our server is behind NAT. externhost = hostname.tld: ( new parameter in Asterisk 1.2.x) externrefresh = xxx: With this parameter we determine how often the DNS entry will be looked up for the hostname defined in the "externhost" parameter (new parameter in Asterisk 1.2.x) = 192.168.1.0/255.255.255.0: definition local network and her masks. fromdomain= : Setting the default domain in the From: field of SIP messages when working as a SIP ua (client). maxexpirey=3600: Maximum allowed registration duration. mineexpiry=60: (Variable introduced since Asterisk 1.4.0) Minimum allowed registration duration. = yes | no (setting for connections with peers) Please note that in Asterisk 1.0.x, the nat parameter can take the following values: no | never | route | yes. notifymimetype=text/plain:Allows you to override the mime type in MWI NOTIFY messages used in messages that are sent by the system. pedantic = yes|no: Enable slow, pedantic checking of Call-ID fields and all lines in the multi-line SIP message header and URI-encoded headers port = : Default SIP port. (this is not the port on which your Asterisk listens for IP packets. See parameter: bindport) promiscredir=yes | no: Enable support for 302 Redirects messages; (No - will forward everything to the local extension that is received in the Contact field, and not to the extension that is specified in the call destination field.) realm = realm (This setting can change the realm for authorization, from the default value - Asterisk, to any selected by you. Works starting from Asterisk version 1.x) register => @ / : Register on the SIP provider’s server. = yes|no: Enable search for tos= calls: This parameter removed starting from Asterisk 1.4.0. Setting IP QoS parameters for outgoing media streams. t1min= : (Variable introduced since Asterisk 1.4.0) The minimum delay for messages to travel to and from the host (roundtrip time) for which the state is being monitored (monitored host). The default value is 100 ms. = : (Variable appeared starting from Asterisk version 1.4.0) Setting IP QoS parameters for SIP packets. = : (Variable appeared starting from Asterisk version 1.4.0) Setting IP QoS parameters for packets transmitting audio data via the RTP protocol. = : (Variable appeared starting from Asterisk version 1.4.0) Setting IP QoS parameters for packets transmitting video data via the RTP protocol.
videosupport=yes|no: Enables SIP video support: The parameter allows you to change the value of the SIP header "User-Agent". = yes|no: If enabled, the resulting Remote-Party-ID value can be trusted.

: The user can be associated with a certain access code (accountcode). Used for . : Allowed and prohibited codecs. : Category for CDR records. You can choose: default, omit, billing, documentation. See: .allowoverlap: Since version 1.4.x. Values: yes/no, default: yes. Setting to "no" disables support for "overlap dialing".allowsubscribe: Since version 1.4.x. Values: yes/no, default: no. Setting to "no" prohibits tracking the status (subscriptions) of this client. allowtransfer: Starting from version 1.4.x. Allow (yes) or deny (no) call transfer for this user.auth: Since version 1.4.x. Link to a record from a section (peer type only).autoframing: Since version 1.4.x. Enable (yes) or disable (no) autoframing"a.buggymwi: Since version 1.4.x. Values: yes/no, default: no. Cisco SIP firmware does not fully support RFC for indicating the presence of a new voicemail (MWI). Set this parameter to "yes" to avoid receiving error messages when sending MWI to phones with this problem: Defines the call group for calls through this device.callingpres: Since version 1.4.x CallerId publishing mode. , see the values ​​in the file doc/callingpres.txt from the distribution asterisk.call-limit: Starting from version 1.4.x__. Limit on the number of calls for the user. For the type "peer" - for outgoing, for "user" - for incoming. .: Enable only if the client supports the function: .cid_number: Since version 1.4.x. Number part of the CallerID.context field: The dial plan context in which calls coming from this client fall: The default IP address of the client when. host= is defined as DYNAMIC. Used if the client has not yet registered, if it is necessary to call it. : How the client handles DMTF signaling. : Specify the username for the "from" field when a call is made using this entry. = : Sets the default domain in the From: field of SIP messages when a call is made using this entry.fullname: Since version 1.4.x. Name part, fields CallerID.g726nonstandard: Since version 1.4.x. Values: yes/no, default: no. If the client is going to “agree” to use the G726-32 audio codec for a communication session, using AAL2 compression, instead of RFC3551 (which is required for devices from Sipura and gateways from Grandstream, and maybe others). This is contrary to the RFC3551 specification, the client _must_ "agree" to use AAL2-G726-32 instead:-(host: How do we find our client - IP address or hostname. If you need the phone to register itself, use the dynamic value instead of the IP address or hostname. : Limit on the number of simultaneous connections with our SIP client. These parameters are obsolete and are not used in version 1.4.x. : very|yes|no|invite|port: Defines how to handle connections to peers. The default value is "no" (authentication is enabled for all connections). (invite and port added in version 1.2.x). : Language code, which are defined in the indications.conf file - where the greeting language and country-specific phone tone settings are defined. : Voice mailbox extension (for the new messages indicator) maxcallbitrate: Starting from version 1.4.x. Maximum bitrate for video in kbit/s. Default: 384 kbit/s.mohsuggest: Since version 1.4.x. See sip.conf for the purpose. : MD5-Hash strings: " :asterisk: " (can be used instead of the secret parameter).musiconhold: Sets the musiconhold class for calls from this SIP phone. For calls that come to this phone, setting this class must be done with a command that is higher in priority (lower priority in the dial plan) than the Dial command in the dial plan. The MusicOnHold parameter is set for calls made, not for phones, and if the command is used to handle a call for an extension, it will override the value specified by this parameter for that call. : This variable changes how the Asterisk server behaves to clients behind a NAT firewall. But this parameter does not solve the problem if Asterisk itself is behind such a firewall, and the client is outside (in internet networks with a normal IP address).outboundproxy: Proxy server address for outgoing calls (peer type only).permit, deny: Restrictions on IP address and IP network for the client. Note: in version 1.4.x the mask parameter is no longer used. : Group for “pickup” calls to your colleagues using the combination “*8” and the Pickup() application to work with the extension *8.port: SIP port of the client.progressinband: Starting from version 1.4.x. Values: yes/no/never, default: never. Should asterisk generate its own ringing tone for caller(in-band ringing). Use the value "never" to never generate this signal, even if some strange devices cannot generate it themselves.promiscredir: Since version 1.4.x. Values: yes/no, default: no. If set to "yes", then we allow SIP messages with code 302 or call forwarding messages to be sent to SIP clients with non-local IP addresses. It is worth noting that the behavior of promiscredir when forwarding a call that is made in local system , can lead to a call loop, because Asterisk cannot handle hairpin calls. : Periodically check client availability. restrictcid: (yes/no) This option is not used, use callingpres instead. rfc2833compensate: Since version 1.4.x. Values: yes/no, default: no. Enabling compensation for the transmission of DTMF signals from other asterisk servers versions up to 1.4.*. In these cases, you must enable this feature or DTMF signal reception will not work properly. : Terminate the call if there is no RTP media stream activity for more than x seconds and if the line is not in hold mode. : Abort the call if there is no RTP media stream activity for more than x seconds, in hold mode (the value must be greater than rtptimeout)secret: If Asterisk acts as a SIP server, then this SIP client must be authorized with this password ( shared secret). If Asterisk acts as a SIP client working with a remote SIP server that requires authorization for "SIP INVITE" requests, then this field will be used for authorization in the "SIP INVITE" messages that our Asterisk will send to the remote SIP server.sendrpid : Whether or not to send the Remote-Party-Id header to this SIP client. Values: yes/no, default: yes.setvar: Since version 1.4.x. The value specifies the "variable=value" pair for the channel variable when calling from this device.subscribecontext: Starting from version 1.4.x. Specifies the name of the context in which SIP requests will be processed SUBSCRIBE.subscribemwi: Starting from version 1.4.x. The values ​​are "yes" and "no". Purpose see sip.conf.t38pt_udptl: Starting from version 1.4.x. Values: yes/no, default: no. Enable/disable T.38 support for this client. : Values: yes/no, default: no. Set to "yes" if you can trust the Remote-Party-ID received from the SIP client. : Client type - outgoing calls only, incoming calls only, or both? useclientcode: Since version 1.4.x. Values: yes/no, default: no.usereqphone: Since version 1.4.x. The values ​​are "yes" and "no". Whether or not to add ";user=phone" to the URI that contains the normal calling number. : If Asterisk is acting as a SIP client working with a remote SIP server that requires authorization for "SIP INVITE" requests, then this field will be used for authorization in the "SIP INVITE" messages that our Asterisk will send to the remote SIP server. And also, for peers that register on our Asterisk server, if this username is used in INVITE messages, as long as it is registered on the server.videosupport: Values: yes/no, default: yes. Option to enable SIP video support. You may need to enable video support for a specific SIP client if, for example, it is disabled in the global part of the configuration, and you want to selectively enable it only for specific users. vmexten: Starting from version 1.4.x. For purpose, see sip.conf (peer type only).

type=friend ; could be: "friend" (this is: peer+user), "peer" or "user"
context=from-sip
username=grandstream1 ; usually, this parameter coincides with the section name -
fromuser=grandstream1 ; reassigns callerid, for example,
; may be necessary for forward calls - FWD
callerid=John Doe
host=192.168.0.23 ; We use a static but intranet IP address
nat=no ; There is no IP address translation (NAT) between our phone and Asterisk
canreinvite=yes ; forces all RTP voice traffic to go through Asterisk
dtmfmode=info ; can be both INBAND and INFO, for example, for BudgeTone phones
mailbox=1234@default ; client has voicebox 1234 in voicebox context: "default"
disallow=all ; you must set disallow=al before using the allow= parameter
allow=ulaw ; Please note: in the client sections, the order in which the codec is selected
; described in the allow= parameter has no meaning!
;allow=alaw
;allow=g723.1 ; Asterisk supports the g723.1 codec only in pass-thru mode!
;allow=g729 ; Only in Pass-thru mode, if you do not have a license to use the g729 codec

Turn off the silence suppression function in the X-Lite client ("Transmit Silence"=YES)!
;Note that Xlite itself sends NAT keep-alive packets, so there is no need for qualify=yes.
type=friend
username=xlite1
callerid="Jane Smith"
host=dynamic
nat=yes ; the X-Lite client is located behind a router with IP address translation (NAT)
canreinvite=no ; Typically set to "NO" if the client is behind NAT
disallow=all
allow=gsm ; The GSM codec requires much less bandwidth, so ulaw.
allow=ulaw
allow=alaw

type=friend
secret=blah
host=dynamic
dtmfmode=inband ; You can choose from: inband, rfc2833, or info
defaultip=192.168.0.59
mailbox=1234,2345 ; Voice box for the "new messages received" indicator
disallow=all
allow=ulaw ; if the mode for dtmf is set to "inband",
allow=alaw ; then we should use only g.711 codec

type=friend
username=user2_pingtel
secret=blah
host=dynamic
qualify=1000 ; We consider the client as non-working,
; if the response from him takes more than 1 second.
callgroup=1,3-4 ; The client is a member of call groups: 1, 3 and 4
pickupgroup=1,3-4 ; We can make "pick-up" calls by pressing *8,
; for calls from groups 1, 3 and 4
defaultip=192.168.0.60
disallow=all
allow=ulaw
allow=alaw
allow=g729

Type=friend
username=user3_cisco
secret=blah
nat=yes ; This phone is behind a router with NAT
host=dynamic
canreinvite=no ; Cisco devices sometimes freeze
; after a certain number of reinvite requests
qualify=200 ; We consider the feast as non-working,
; if the response from it comes longer than 200ms
defaultip=192.168.0.4
disallow=all
allow=ulaw
allow=alaw
allow=g729

type=friend

Username=user4_cisco
fromuser=markster ; Define the username to be placed in the "from" field instead of callerid
secret=blah
defaultip=192.168.0.4 ; use either host=dynamic or defaultip=...
amaflags=default ; Could be: default, omit, billing, documentation
accountcode=markster ; The user is associated with an accountcode to simplify billing.
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1

The description below is taken from the Digium Asterisk handbook.

After you define these SIP accounts for your clients in the SIP.conf file, then these clients have the opportunity to register on the asterisk server and make outgoing calls. In order for them to be able to receive calls, you need to describe extensions for them in your dial plan file. Example:

Exten => 1010,1, Dial(SIP/user3_cisco,10,t)

If someone makes a call to 1010, then an attempt will be made to call the SIP client registered as user3_cisco.

[fwd.pulver.com]
type=friend
secret=
username=
host=fwd.pulver.com
insecure=very ; required to support incoming FWD calls

Now let's go to the file description and see how we can configure contexts in terms of dialing.

[Provider]
type=friend
username=XXXXX
secret=XXXXX
host=xxxxx.FakeProvider.com

So now when Asterisk receives a call with SER, it performs the "autocreatepeer" action and allows the call to be processed in the OUTGOING context.