Asterisk compile error solution “‘pjsip_tcp_transport_cfg’ has no member named ‘sockopt_params’”

Once I compiled Asterisk version 13.13.1 and when running make I noticed the following error:

‘pjsip_tcp_transport_cfg’ has no member named ‘sockopt_params’

pjproject-2.2.1 has already been compiled.

Solved the problem by compiling a newer version of pjproject-2.4.5

cd /usr/src
wget http://www.pjsip.org/release/2.4.5/pjproject-2.4.5.tar.bz2
tar -xjvf pjproject-2.4.5.tar.bz2
cd pjproject-2.4.5
CFLAGS='-DPJ_HAS_IPV6=1' ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr
make dep
make
make install

After that, the error disappeared.

Configuring Fail2Ban for Asterisk

On the test I will use Asterisk 13.1.0 and Fail2Ban 0.9.3-1 installed in Ubuntu Server 16.04.1 LTS.

Install Fail2Ban as I wrote in this article – Installing and Configuring Fail2ban

Open the configuration file Asterisk responsible for logging events in /var/log/asterisk/messages:

sudo nano /etc/asterisk/logger.conf

Add security to messages:

messages => notice,warning,error,security

Restart the asterisk logging system:

sudo asterisk -rvv
logger reload
quit

Add the Asterisk configuration file to the directory with the Fail2Ban configuration, thus activating the monitoring of its logs:

sudo nano /etc/fail2ban/jail.d/asterisk.conf

where 86400 in seconds = 24 hours, that is, the attacker will be blocked for a day.

[asterisk]
enabled = true
bantime = 86400

Or, change the file /etc/fail2ban/jail.conf where [asterisk-tcp] and [asterisk-udp] are false to true.

Restart fail2ban for the new configuration file to load:

sudo fail2ban-client reload

Let’s check the work:

sudo fail2ban-client status asterisk

Done, now Fail2Ban will block IP addresses from which the passwords to Asterisk accounts are not correctly entered.

Sending Asterisk voicemail to multiple emails

Let’s say the voice mail is configured as I described in the article – Setting up voicemail in Asterisk.
There is the following context:

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

If you want to send a voice message to several email addresses, then instead of test@example.com, for example, testmail:

[voicemailcontext]
207 => 1111,Username,testmail,,attach=yes|tz=ua|delete=yes

Then open the /etc/aliases file in a text editor:

sudo nano /etc/aliases

And let’s specify aliases for testmail:

testmail: support@example.net,user@example.net

For the changes to take effect, you need to update the alias database with the command:

cd /etc
sudo newaliases

Done, Asterisk will send a message to testmail, and it will be automatically forwarded to the specified addresses.

See also:
Redirecting mail for the root user

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