Difference between revisions of "Asterisk"

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[[Category:Server]]
+
[[Category:Telephony]]
=Asterisk Configration=
+
[[Category:Voice over IP]]
==Summary==
+
[[ja:Asterisk]]
Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
+
[http://www.asterisk.org Asterisk] is a complete PBX (private branch exchange) in software. It runs on Linux, BSD, Windows and macOS and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
  
Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny.  
+
Asterisk provides voice-mail services with directory, call conferencing, interactive voice response and call queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny.  
  
This wiki will show you how to configure a simple in house network enabling us to use a SIP softphone to talk to another SIP softphone on your LAN. This is nothing major, but will get you familiar with Asterisk configuration and its pretty cool.  
+
This article will show you how to configure a simple in house network enabling us to use a SIP [[softphone]] to talk to another SIP softphone on your LAN.
  
==What You Need==
+
==Installation==
Of course, youll need [http://www.asterisk.org Asterisk], a SIP softphone and at least two machines. Recommendations for SIP phones are [http://www.wirlab.net/kphone/ kphone] and [http://www.xten.net/index.php?menu=X-Series x-lite].
+
 
 +
Install the {{AUR|asterisk}} package. If you are using Cisco-based phones it is recommended to use the {{AUR|asterisk-cisco}} package instead as this is prepatched with the presence patch. (See [https://issues.asterisk.org/jira/browse/ASTERISK-13145 https://issues.asterisk.org/jira/browse/ASTERISK-13145])
 +
 
 +
[[Start]] the server with {{ic|asterisk.service}}.
 +
 
 +
You will also need a SIP [[softphone]] and at least two machines. Recommendations for SIP phones are [http://www.linphone.org/ Linphone] ({{AUR|linphone}}{{Broken package link|package not found}} package) or [http://www.counterpath.com/x-lite/ X-Lite] ({{AUR|xlite_bin}} package).
 +
 
 +
To enable ilbc codec support add the following to the very beginning of the {{ic|build}} section of the PKGBUILD:
 +
{{bc|<nowiki>
 +
cd ${srcdir}/${pkgname}-${pkgver}/contrib/scripts
 +
echo | ./get_ilbc_source.sh
 +
</nowiki>}}
  
 
==Configuration==
 
==Configuration==
Assuming your asterisk server is up and running we will only need to edit two files sip.conf and extensions.conf.  
+
===SIP===
Change to your asterisk configuration directory (should be <code>/etc/asterisk</code>). Edit sip.conf and place the following in:
+
Assuming your asterisk server is up and running, we will only need to edit two files: {{ic|sip.conf}} and {{ic|extensions.conf}}. Change to your asterisk configuration directory (should be {{ic|/etc/asterisk}}). Edit sip.conf and place the following:
<pre>
+
{{bc|1=
 
[me1]
 
[me1]
 
type=friend
 
type=friend
 
username=me1
 
username=me1
secret=password
+
secret=PASSWORD
 
host=dynamic
 
host=dynamic
 
context=house
 
context=house
Line 25: Line 36:
 
type=friend
 
type=friend
 
username=me2
 
username=me2
secret=password
+
secret=PASSWORD
 
host=dynamic
 
host=dynamic
 
context=house
 
context=house
</pre>
+
}}
  
This creates our two SIP users <code>me1</code> and <code>me2</code> with a password of <code>password</code> in the <code>house</code> context. We will be defining the context next.
+
This creates our two SIP users {{Ic|me1}} and {{Ic|me2}} with a password of {{Ic|PASSWORD}} in the {{Ic|house}} context.
  
Edit extensions.conf with the following:
+
We will be defining the context next -- edit extensions.conf with the following:
<pre>
+
{{bc|1=
 
[house]
 
[house]
 
exten => 100,1,Dial(SIP/me1)
 
exten => 100,1,Dial(SIP/me1)
  
 
exten => 101,1,Dial(SIP/me2)
 
exten => 101,1,Dial(SIP/me2)
</pre>
+
}}
 +
 
 +
This creates the context {{Ic|house}} and assigns extension 100 to the SIP user {{Ic|me1}}, and extension 101 to the SIP user {{Ic|me2}}. Now all thats left is to see if it works.
 +
===Music on hold===
 +
Music on hold is a really sweet feature. And once again easy to install and configure.
 +
Edit {{ic|/etc/asterisk/musiconhold.conf}} and add, or make sure it is uncommented:
 +
{{bc|1=
 +
[default]
 +
mode=files
 +
directory=mohmp3
 +
}}
 +
 
 +
And that is all there is to it. Just copy your favorite legally obtained MP3 to {{ic|/var/lib/asterisk/mohmp3}}.
 +
 
 +
===Voicemail===
 +
Voicemail is another feature of asterisk. There are many ways to configure it, however this article only covers a simple approach.
 +
 
 +
Create/edit your {{ic|voicemail.conf}}:
 +
{{bc|<nowiki>
 +
[general]
 +
format=gsm|wav49|wav
 +
serveremail=asterisk
 +
attach=no
 +
mailcmd=/usr/sbin/sendmail -t
 +
maxmessage=180
 +
maxgreet=60
 +
 
 +
[default]
 +
100 => 1234,Me,me@mydomain.com
 +
</nowiki>}}
 +
 
 +
What does this mean? Most of the {{Ic|[general]}} is pretty self-explanatory. However, do note that if you have postfix set up right the PBX will send an email notifying the user of a new voice-mail and if {{Ic|1=attach=yes}} is defined it will attach the file.
  
This creates that context <code>house</code> and assigns extension 100 to the SIP user <code>me1</code>, and extension 101 to the SIP user <code>me2</code>. Now all thats left is to see if it works.
+
Now for the actual mailbox. The format is:
 +
mailbox => password,user,email
 +
In this case, we gave 'Me' (email me@mydomain.com) mailbox 100, with a password of 1234.
  
==Asterisk Console And Softphones==
+
Now we have to have a way to leave messages to this voice-mail, and a way to access it.
Now lets get Asterisk going. To do this enter
+
For this, we go back to the {{ic|extensions.conf}} and modify your existing entry as follows:
<pre>
+
exten => 100,1,Dial(SIP/me1,20)
# asterisk -vvvvvvc
+
exten => 100,n,Voicemail(u100@default)
</pre>
+
The 20 on the end of the first 'exten' tells 'Dial()' to call for 20 seconds. If no one answers it heads to voice-mail box 100 in the default context.
This will give us the Asterisk CLI with verbose output. If Asterisk is already running you'll need to use
+
 
<pre>
+
Next is actually accessing your voicemail. For this we add:
# asterisk -r
+
exten => 600,1,VoiceMailMain,s100@default
</pre>
+
So when we call 600, the application 'VoiceMailMain' goes to 100 in the default context. The {{Ic|s}} allows for automatic login.
Now fire up your SIP clients and set them up with the information in the sip.conf. Switch back to your Asterisk CLI and you should see  
+
 
<pre>
+
{{Note|The 'VoiceMail' applications have a significant amount of options, so it is suggested reading over some additional documentation. This is just for a basic, home use setup.  Also note that it is generally a good idea to use extensions higher then your users extensions for accessing 'VoiceMail'. This way someone dialing 208 does not hit someone's voice-mail at 205.}}
 +
 
 +
===Connecting to the PSTN===
 +
Now that you have the previous setup, it is time to actually connect to the outside world. To do this, you will need a provider such as [https://www.onsip.com/ OnSIP]. Your provider should have instructions on connecting to asterisk, so this section is very general.
 +
 
 +
====General set-up====
 +
=====sip.conf=====
 +
{{bc|1=
 +
[general]
 +
register => username:password@sip.specific.com
 +
 
 +
[whatever]                 
 +
fromdomain=specific.com   
 +
host=sip.specific.com
 +
insecure=very    ; check with provider
 +
username=usernameduh
 +
secret=passwordduh
 +
type=peer
 +
}}
 +
 
 +
=====extensions.conf=====
 +
{{bc|1=
 +
[outboundwithCID]  ; this can be whatever
 +
exten => _1NXXNXXXXXX,1,SetCIDNum(15555551234)
 +
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@whatever)
 +
exten => _1NXXNXXXXXX,3,Congestion()
 +
exten => _1NXXNXXXXXX,103,Busy()
 +
 
 +
[default]  ; This should be set in your sip.conf for incoming calls
 +
 
 +
;These should to be changed to your actual number
 +
; ie    15555555555
 +
exten => 1NXXNXXXXXX,1,Answer()
 +
exten => 1NXXNXXXXXX,2,Playback(ttt-weasels)
 +
exten => 1NXXNXXXXXX,3,HangUp()
 +
}}
 +
 
 +
*In the outbound context, any number dialed will be sent out to your service provider. The 'whatever' in the 2 priority should match what you have in your sip.conf.
 +
*Of course, the inbound dial-plan can be modified to do what you want. For instance, you can have {{Ic|Dial(SIP/me1)}} so when someone calls your number they are routed to your SIP phone on your computer. Then add in voice-mail and so on.
 +
 
 +
=====iax.conf=====
 +
The first step is to log into FWD and enable their side of IAX. It is under extra features, and keep in mind that the authors claim it takes a little while to activate.
 +
 
 +
Now edit your iax.conf with the following in the 'general' section:
 +
{{bc|1=
 +
register => FWDNUMBER:PASSWORD@iax2.fwdnet.net
 +
disallow = all
 +
allow = ulaw
 +
}}
 +
 
 +
And at the bottom add:
 +
{{bc|1=
 +
[iaxfwd]
 +
type=user
 +
context=fromiaxfwd
 +
auth=rsa
 +
inkeys=freeworlddialup
 +
}}
 +
 
 +
This allows calls from FWD.
 +
 
 +
=====extensions.conf=====
 +
Place this at the top under '[globals]':
 +
{{bc|1=
 +
FWDNUMBER=MYFWDNUMBER ; your calling number
 +
FWDCIDNAME="MyName"; your caller id
 +
FWDPASSWORD=MYFWDPASSWORD ; your password
 +
FWDRINGS=sip/office ; the phone to ring
 +
FWDVMBOX=1000 ; the VM box for this user
 +
}}
 +
 
 +
Next, add this to a context for outgoing:
 +
{{bc|1=
 +
exten => _393.,1,SetCallerId,${FWDCIDNAME}
 +
exten => _393.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r)
 +
exten => _393.,3,Congestion
 +
}}
 +
 
 +
You can change the '393' to whatever you want. This is what you will dial before dialing a 'fwd' number. For instance, to dial '744561' you would dial '393744561'.
 +
 
 +
And lastly, the incoming calls:
 +
{{bc|1=
 +
[fromiaxfwd]
 +
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
 +
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
 +
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
 +
}}
 +
 
 +
{{Note|If you have problems try removing the variables from {{ic|extensions.conf}}. These instructions are from FWD's site and I have not been tested by this article's author.}}
 +
 
 +
Extensions to try calling are 55555 (a volunteer maned test line) and 514 (conference).
 +
 
 +
===Sounds===
 +
Sounds are stored in the folder {{ic|/var/lib/asterisk/xx}}, {{ic|xx}} stands for the code of the language for example "en" for English. To add new sounds copy them to the folder. Preserve the following folder structure:
 +
{{bc|
 +
/var/lib/asterisk/sounds/xx
 +
/var/lib/asterisk/sounds/xx/digits
 +
/var/lib/asterisk/sounds/xx/letters
 +
/var/lib/asterisk/sounds/xx/phonetic
 +
}}
 +
 
 +
Edit the language parameter in the {{ic|sip.conf}}
 +
{{bc|1=
 +
[general]
 +
...
 +
language=en
 +
...
 +
}}
 +
Possible sources for sounds are:
 +
* http://downloads.asterisk.org/pub/telephony/sounds/
 +
* https://packages.debian.org/wheezy/all/asterisk-prompt-xx
 +
** [https://packages.debian.org/wheezy/all/asterisk-prompt-fr fr]
 +
** [https://packages.debian.org/wheezy/all/asterisk-prompt-de de]
 +
** ...
 +
* [http://www.voip-info.org/wiki/view/Asterisk+sound+files+international voip-info.org]
 +
===MeetMe===
 +
MeetMe is the application that allows you to do conference calling. Same as everything, basic setup is easy.
 +
 
 +
Edit {{ic|meetme.conf}}:
 +
conf => 1000
 +
Next is extensions.conf
 +
exten => 999,1,MeetMe(1000|M)
 +
Now dial 999 to get into conference 1000. The {{Ic||M}} enables music on hold if no one is in there. It will automatically go away when someone joins the conference.
 +
 
 +
{{Note|You ''must'' have the zaptel package in order for MeetMe to work. Install it and run {{Ic|modprobe ztdummy}} before running asterisk. This provides digium timing for us without cards so we can utilize TDM.}}
 +
 
 +
==Asterisk console and softphones==
 +
Now lets get Asterisk going:
 +
# asterisk -vvvvvvc
 +
This will give us the Asterisk CLI with verbose output. If Asterisk is already running you will need to use:
 +
# asterisk -r
 +
Now fire up your SIP clients and set them up with the information in the sip.conf. Switch back to your Asterisk CLI and you should see:
 
  Registered SIP 'me1' at 192.168.0.142 port 5061 expires 60
 
  Registered SIP 'me1' at 192.168.0.142 port 5061 expires 60
</pre>
 
 
Now you should be able to dial <code>101</code> from <code>me1</code> and talk to me2.
 
  
==Common Problems==
+
Now you should be able to dial {{Ic|101}} from {{Ic|me1}} and talk to {{Ic|me2}}.
If you recieve a 404 Not Found error check your extensions.conf and the number you dialed.
 
  
==Coming Up==
+
==Troubleshooting==
*Voicemail
+
If you receive a 404 Not Found error check your {{ic|extensions.conf}} and the number you dialed.
*Connecting To The Outside
 
*Music On Hold
 

Latest revision as of 12:26, 30 November 2019

Asterisk is a complete PBX (private branch exchange) in software. It runs on Linux, BSD, Windows and macOS and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides voice-mail services with directory, call conferencing, interactive voice response and call queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny.

This article will show you how to configure a simple in house network enabling us to use a SIP softphone to talk to another SIP softphone on your LAN.

Installation

Install the asteriskAUR package. If you are using Cisco-based phones it is recommended to use the asterisk-ciscoAUR package instead as this is prepatched with the presence patch. (See https://issues.asterisk.org/jira/browse/ASTERISK-13145)

Start the server with asterisk.service.

You will also need a SIP softphone and at least two machines. Recommendations for SIP phones are Linphone (linphoneAUR[broken link: package not found] package) or X-Lite (xlite_binAUR package).

To enable ilbc codec support add the following to the very beginning of the build section of the PKGBUILD:

cd ${srcdir}/${pkgname}-${pkgver}/contrib/scripts
echo | ./get_ilbc_source.sh

Configuration

SIP

Assuming your asterisk server is up and running, we will only need to edit two files: sip.conf and extensions.conf. Change to your asterisk configuration directory (should be /etc/asterisk). Edit sip.conf and place the following:

[me1]
type=friend
username=me1
secret=PASSWORD
host=dynamic
context=house

[me2]
type=friend
username=me2
secret=PASSWORD
host=dynamic
context=house

This creates our two SIP users me1 and me2 with a password of PASSWORD in the house context.

We will be defining the context next -- edit extensions.conf with the following:

[house]
exten => 100,1,Dial(SIP/me1)

exten => 101,1,Dial(SIP/me2)

This creates the context house and assigns extension 100 to the SIP user me1, and extension 101 to the SIP user me2. Now all thats left is to see if it works.

Music on hold

Music on hold is a really sweet feature. And once again easy to install and configure. Edit /etc/asterisk/musiconhold.conf and add, or make sure it is uncommented:

[default]
mode=files
directory=mohmp3

And that is all there is to it. Just copy your favorite legally obtained MP3 to /var/lib/asterisk/mohmp3.

Voicemail

Voicemail is another feature of asterisk. There are many ways to configure it, however this article only covers a simple approach.

Create/edit your voicemail.conf:

[general]
format=gsm|wav49|wav
serveremail=asterisk
attach=no
mailcmd=/usr/sbin/sendmail -t
maxmessage=180
maxgreet=60

[default]
100 => 1234,Me,me@mydomain.com

What does this mean? Most of the [general] is pretty self-explanatory. However, do note that if you have postfix set up right the PBX will send an email notifying the user of a new voice-mail and if attach=yes is defined it will attach the file.

Now for the actual mailbox. The format is:

mailbox => password,user,email

In this case, we gave 'Me' (email me@mydomain.com) mailbox 100, with a password of 1234.

Now we have to have a way to leave messages to this voice-mail, and a way to access it. For this, we go back to the extensions.conf and modify your existing entry as follows:

exten => 100,1,Dial(SIP/me1,20)
exten => 100,n,Voicemail(u100@default)

The 20 on the end of the first 'exten' tells 'Dial()' to call for 20 seconds. If no one answers it heads to voice-mail box 100 in the default context.

Next is actually accessing your voicemail. For this we add:

exten => 600,1,VoiceMailMain,s100@default

So when we call 600, the application 'VoiceMailMain' goes to 100 in the default context. The s allows for automatic login.

Note: The 'VoiceMail' applications have a significant amount of options, so it is suggested reading over some additional documentation. This is just for a basic, home use setup. Also note that it is generally a good idea to use extensions higher then your users extensions for accessing 'VoiceMail'. This way someone dialing 208 does not hit someone's voice-mail at 205.

Connecting to the PSTN

Now that you have the previous setup, it is time to actually connect to the outside world. To do this, you will need a provider such as OnSIP. Your provider should have instructions on connecting to asterisk, so this section is very general.

General set-up

sip.conf
[general]
register => username:password@sip.specific.com

[whatever]                   
fromdomain=specific.com     
host=sip.specific.com
insecure=very    ; check with provider
username=usernameduh
secret=passwordduh
type=peer
extensions.conf
[outboundwithCID]  ; this can be whatever
exten => _1NXXNXXXXXX,1,SetCIDNum(15555551234)
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@whatever)
exten => _1NXXNXXXXXX,3,Congestion()
exten => _1NXXNXXXXXX,103,Busy()

[default]  ; This should be set in your sip.conf for incoming calls

;These should to be changed to your actual number
; ie     15555555555
exten => 1NXXNXXXXXX,1,Answer()
exten => 1NXXNXXXXXX,2,Playback(ttt-weasels)
exten => 1NXXNXXXXXX,3,HangUp()
  • In the outbound context, any number dialed will be sent out to your service provider. The 'whatever' in the 2 priority should match what you have in your sip.conf.
  • Of course, the inbound dial-plan can be modified to do what you want. For instance, you can have Dial(SIP/me1) so when someone calls your number they are routed to your SIP phone on your computer. Then add in voice-mail and so on.
iax.conf

The first step is to log into FWD and enable their side of IAX. It is under extra features, and keep in mind that the authors claim it takes a little while to activate.

Now edit your iax.conf with the following in the 'general' section:

register => FWDNUMBER:PASSWORD@iax2.fwdnet.net 
disallow = all
allow = ulaw

And at the bottom add:

[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup

This allows calls from FWD.

extensions.conf

Place this at the top under '[globals]':

FWDNUMBER=MYFWDNUMBER ; your calling number
FWDCIDNAME="MyName"; your caller id
FWDPASSWORD=MYFWDPASSWORD ; your password
FWDRINGS=sip/office ; the phone to ring
FWDVMBOX=1000 ; the VM box for this user

Next, add this to a context for outgoing:

exten => _393.,1,SetCallerId,${FWDCIDNAME}
exten => _393.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r)
exten => _393.,3,Congestion

You can change the '393' to whatever you want. This is what you will dial before dialing a 'fwd' number. For instance, to dial '744561' you would dial '393744561'.

And lastly, the incoming calls:

[fromiaxfwd]
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
Note: If you have problems try removing the variables from extensions.conf. These instructions are from FWD's site and I have not been tested by this article's author.

Extensions to try calling are 55555 (a volunteer maned test line) and 514 (conference).

Sounds

Sounds are stored in the folder /var/lib/asterisk/xx, xx stands for the code of the language for example "en" for English. To add new sounds copy them to the folder. Preserve the following folder structure:

/var/lib/asterisk/sounds/xx
/var/lib/asterisk/sounds/xx/digits
/var/lib/asterisk/sounds/xx/letters
/var/lib/asterisk/sounds/xx/phonetic

Edit the language parameter in the sip.conf

[general]
...
language=en
...

Possible sources for sounds are:

MeetMe

MeetMe is the application that allows you to do conference calling. Same as everything, basic setup is easy.

Edit meetme.conf:

conf => 1000

Next is extensions.conf

exten => 999,1,MeetMe(1000|M)

Now dial 999 to get into conference 1000. The enables music on hold if no one is in there. It will automatically go away when someone joins the conference.

Note: You must have the zaptel package in order for MeetMe to work. Install it and run modprobe ztdummy before running asterisk. This provides digium timing for us without cards so we can utilize TDM.

Asterisk console and softphones

Now lets get Asterisk going:

# asterisk -vvvvvvc

This will give us the Asterisk CLI with verbose output. If Asterisk is already running you will need to use:

# asterisk -r

Now fire up your SIP clients and set them up with the information in the sip.conf. Switch back to your Asterisk CLI and you should see:

Registered SIP 'me1' at 192.168.0.142 port 5061 expires 60

Now you should be able to dial 101 from me1 and talk to me2.

Troubleshooting

If you receive a 404 Not Found error check your extensions.conf and the number you dialed.