Setting up voicemail in Asterisk

For example, I’ll set up voice mail for SIP number 207.
Voice messages will be sent to the email using Postfix.
How to install it I described in this article – Installing and Configuring Postfix.

For starters, let’s point out the following in the context of SIP 207 (usually in /etc/asterisk/sip.conf):

mailbox=207@voicemailcontext

Next, configure the configuration of voice mail in the file /etc/asterisk/voicemail.conf:

[general]
; Format of audio files
format=wav49|gsm|wav
; From whom to send letters with notifications
serveremail=noreply@example.com
; Whether to attach to an audio file
attach=yes
; The maximum number of messages (standard 100, maximum 9999)
maxmsg=100
; Maximum message time in seconds
maxsecs=120
; Maximum greeting time in seconds
maxgreet=60
; Number of seconds of silence before recording is complete
maxsilence=10
; Threshold sensitivity to silence, the lower the sensitivity, the value from 0 to 256, standard 128
silencethreshold=128
; Maximum number of failed connection attempts
maxlogins=3
; Automatically move the listened messages to the "Old" folder. The default is on.
moveheard=yes
; The encoding of messages, the standard ISO-8859-1, with it my part of the text was displayed incorrectly, so it's better to specify UTF-8
charset=UTF-8
; Skip the line "[PBX]:" from the message header
pbxskip=yes
; The text of the line "From:"
fromstring=VoiceMail
; Letter subject
emailsubject=New voice message ${VM_MSGNUM} in the mailbox ${VM_MAILBOX}
; The contents of the letter
emailbody=Dear ${VM_NAME}:\n\n\tYou received a new voice message in length ${VM_DUR} under the number (number ${VM_MSGNUM})\nin the mailbox ${VM_MAILBOX} from ${VM_CALLERID}, at ${VM_DATE}. \n\t
; Date format
emaildateformat=%A, %d %B %Y в %H:%M:%S
pagerdateformat=%T %D
; Standard program for sending mail
mailcmd=/usr/sbin/sendmail -t

[zonemessages]
ru=Europe/Moscow|'vm-received' q 'digits/at' H 'hours' M 'minutes'
ua=Europe/Kiev|'vm-received' q 'digits/at' H 'hours' M 'minutes'

; We will write the context parameters voicemailcontext, 1111 - voice mail password (you can not specify), Username - user name, test@example.com - which address to send voice messages, after the comma you can specify one more, at the end of the option
[voicemailcontext]
207 => 1111,Username,test@example.com,,attach=yes|tz=ua|delete=yes

By the way, if you do not specify “delete=yes”, when the maxmsg limit is reached, the answering machine will say a greeting, and then the text that the subscriber’s voice box is full and do not save the message, or send it to the email. In this case, you need to call the voicemail number and delete the messages. If “delete=yes” is specified, the messages are not stored on the server, they do not come to the voice mailbox, but only sent to the email, in this case the maxmsg limit does not work and the overflow is not possible.

Now, in the configuration of the dial plan /etc/asterisk/extensions.conf in the main context, add the number by calling to which you can listen to the mail:

exten => 500,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to VoiceMail (500))
exten => 500,n,VoiceMailMain(0${CALLERID(num)}@voicemailcontext,s)
exten => 500,n, Hangup

And add the VoiceMail line to the dialing context of the number 207 (after which, if the number is not answered or not on the network, voice mail will work), for example:

[207]
exten => 207,1,Dial(SIP/207,30)
exten => 207,n,Answer
exten => 207,n,VoiceMail(207@voicemailcontext)

Finally we connect to the Asterisk console, reload the configuration, see the list of voice mailboxes and letters:

asterisk -rvv
sip reload
voicemail reload
dialplan reload
voicemail show users
exit

The recorded messages are stored in the directory /var/spool/asterisk/voicemail/
Sound files are stored in /usr/share/asterisk/sounds

See also:
Sending Asterisk voicemail to multiple emails
Solving the error in Asterisk “File vm-newn does not exist in any format”
How to convert audio files to ulaw, alaw, gsm, g722, etc. for Asterisk

Asterisk time based routing

Here is an example of routing calls over time in Asterisk.
Assume that the /etc/asterisk/extensions.conf file has a configured context for the trunk with the following parameters:

[Trunk_2]
exten => s,1,DIAL(SIP/6004&SIP/6003,19)
exten => s,2,Hangup()

And to resolve the time calls to different phones, we will point out for the context of this trunk only nested contexts:

[Trunk_2]
include => daytime,8:00-18:00,mon-sat,*,*
include => nighttime,18:00-8:00,mon-sun,*,*
include => sunday,8:00-22:00,sun,*,*

In fact, daytime, nighttime, sunday are only the names of contexts for which the time is written in the context of [Trunk_2], they can be called anything.

And then in these separate contexts we will already add the necessary extensions.
That is, in the afternoon:

[daytime]
exten => s,1,DIAL(SIP/6004&SIP/6003,19)
exten => s,2,Hangup()

At night:

[nighttime]
exten => s,1,DIAL(SIP/6002,19)
exten => s,2,Hangup()

And on Sunday:

[sunday]
exten => s,1,DIAL(SIP/6002,19)
exten => s,2,Hangup()

You can also, for example, create a holiday context with holidays:

include => holiday,*,*,1,jan
include => holiday,*,*,8,mar

etc.

For those who want to paint more in detail by day, I’ll give you a list of days in English:
mon – Monday
tue – Tuesday
wed – Wednesday
thu – Thursday
fri – Friday
sat – Saturday
sun – Sunday

Similarly, up to three letters and names of months are shortened.

Configuring Automatic Calls in Asterisk

Asterisk can automatically make a call if you put a .call file in the (default) /var/spool/asterisk/outgoing/ directory. If the date of the file change is greater than the current one, the call will be made on or after this time.

For automatic calls, the pbx_spool.so module must be loaded, it must be registered in modules.conf or autoload=yes must be specified.

Continue reading “Configuring Automatic Calls in Asterisk”

Asterisk warning “leave_voicemail: No more messages possible”

I noticed the following error on one of the servers:

WARNING[21992][C-00000b27]: app_voicemail.c:6559 leave_voicemail: No more messages possible

It turned out that the mailbox was full of voice messages and they ceased to exist, in response the caller was informed “The subscriber’s voice box is full”.

To solve this problem there are several options:

1) Delete the messages in the voice mailbox by calling the voice mail number.

2) Increase the value of maxmsg in the voicemail.conf file, thereby increasing the maximum number of messages in the mailbox, but again it may be full. After the changes in the voicemail.conf file, you need to apply them:

sudo asterisk -rvv
voicemail reload
quit

3) In the context of the voice mailbox, add delete=yes, for example:

[voicemailcontext]
207 => 1111,Username,test@example.com,,attach=yes|tz=ua|delete=yes

In this case, voice messages will be sent to e-mail, and they will be immediately deleted from the server, that is, they can not be listened to by calling to the voice mail number and accordingly the mailbox will never be full. I consider this option the best.

See also:
Setting up voicemail in Asterisk

Adding a SIP client to FreePBX

To add a SIP client to FreePBX, open the menu “Applications” – “Extensions“, choose for example “Generic CHAN SIP Device” and we indicate the main parameters:

User Extension: 6000 (SIP number)
Display Name: Operator (any name to display)
Secret: PASSWORD
and click “Submit“.

Done, SIP is added, it can already be registered at the specified number and password.

How to add SIP in the configuration file I described in this article – Adding SIP clients to Asterisk

Installing the Digium Asterisk GUI

Digium Asterisk GUI – web-management interface Asterisk.

Today I’ll sculpt it to Asterisk 11 on Ubuntu Server 14.04 LTS.
Switch directly to the root user:

sudo -i

Download it:

apt-get install subversion
mkdir -p ~/asterisk-gui
cd ~/asterisk-gui
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0/

We compile and install:

cd 2.0
./configure
make
make install

Just in case, we’ll make a copy of the Asterisk configuration files:

cp -r /etc/asterisk /etc/asterisk.original

Open the configuration file manager.conf for example in the editor nano (Ctrl+X to exit the editor, y/n to save or cancel changes):

nano /etc/asterisk/manager.conf

The main parameters that must be configured in the manager.conf configuration file are:

[general]
enabled = yes
webenabled = yes
bindaddr = 0.0.0.0
[USERNAME]
secret = PASSWORD
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config

At the very end of the file, replace the symbol # with; otherwise you can not enter under the login and password specified above.

Now edit http.conf:

nano /etc/asterisk/http.conf

In it we will specify the following parameters of the web server:

enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
prefix=asterisk

Let’s check the settings:

make checkconfig

Delete the empty folder and specify a link to the real one with the files of the web server:

rmdir /usr/share/asterisk/static-http/
ln -s /var/lib/asterisk/static-http/ /usr/share/asterisk/

Restarting Asterisk:

/etc/init.d/asterisk restart

Now Digium Asterisk GUI should open by link http://192.168.56.102:8088/asterisk/static/config/index.html, where 192.168.56.102 this is the IP or Server domain with Asterisk.

To log in, we’ll specify the username and password you wrote earlier in the manager.conf file

Done.