Setting up Asterisk on Ubuntu Bionic

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Requirements

SIP is time sensitive

systemctl status ntp #should not run
apt -y install ntpdate
ntpdate -u 0.ru.pool.ntp.org
hwclock --systohc

apt -y install ntp
systemctl status ntp

chose the nearest servers and listen on your subnet

vi /etc/ntp.conf

server 0.md.pool.ntp.org
server 1.europe.pool.ntp.org
server 3.europe.pool.ntp.org

restrict x.x.x.0 mask 255.255.255.0

systemctl restart ntp
ntpq -p

jansson libraries need to be >= 2.11, hence Xenial does not do (2.7). Bionic and Focal are ok.

lsb_release -a
apt -y install build-essential subversion libncurses5-dev libssl-dev libxml2-dev
apt -y install wget libnewt-dev linux-headers-`uname -r`
apt -y install libedit-dev uuid-dev libjansson-dev libsqlite3-dev
apt -y install libsrtp2-dev

Build & Install

seek for the latest version

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
tar xzf asterisk-18-current.tar.gz
cd asterisk-18.1.0/
./configure
#./configure NOISY_BUILD=yes

should get

grep PBX_SRTP= config.log
grep HAVE_SRTP config.log

build

#make menuconfig
grep ^proc /proc/cpuinfo
time make -j12 > ../asterisk.log && echo BUILT
tail -20 ../asterisk.log
make install

install the configuration samples

make samples
#make basic-pbx
ls -lF /etc/asterisk/

avoid running it as root (eventually choose a specific UID/GID for it)

useradd -s /bin/false -d /var/lib/asterisk asterisk
chown -R asterisk:asterisk /var/spool/asterisk /var/lib/asterisk /var/run/asterisk /var/log/asterisk/cdr-csv

deploy some init script

cp contrib/init.d/rc.debian.asterisk /etc/init.d/asterisk
vi /etc/init.d/asterisk

DAEMON=/usr/sbin/asterisk
ASTVARRUNDIR=/var/run/asterisk
ASTETCDIR=/etc/asterisk

cp contrib/init.d/etc_default_asterisk /etc/default/asterisk
vi /etc/default/asterisk

AST_USER="asterisk"
AST_GROUP="asterisk"

fortunately, that still works on Bionic and Focal

update-rc.d asterisk defaults

Ready to go

Check the path and version

which asterisk
asterisk -V
#asterisk -cvvv

Run the daemon

/etc/init.d/asterisk start
/etc/init.d/asterisk status
#/etc/init.d/asterisk stop
ps auxfww | grep ^asterisk
netstat -lntup | grep asterisk

SIP Peers and Dial Plan

warning: using version 15 with chan_sip for now

TODO switch to pjsip

cd /etc/asterisk/

prepare for device registration

mv sip.conf sip.conf.dist
sed '/^[[:space:]]*;/d; /^$/d' sip.conf.dist > sip.conf
vi sip.conf

[general]
...
qualify=yes

[testphone]
        type=friend
        context=phones
        allow=ulaw,alaw
        secret=PASS
        host=dynamic

[testphone2]
        type=friend
        context=phones
        allow=ulaw,alaw
        secret=PASS2
        host=dynamic

mv extensions.conf extensions.conf.dist
sed '/^[[:space:]]*;/d; /^$/d' extensions.conf.dist > extensions.conf
vi extensions.conf

[phones]

exten => 101,1,NoOp(testphone is being called)
exten => 101,2,Dial(SIP/testphone)
exten => 101,3,Hangup

exten => 102,1,NoOp(testphone2 is being called)
exten => 102,2,Dial(SIP/testphone2)
exten => 102,3,Hangup

Get to the CLI with verbosity level 3, and apply,

/etc/init.d/asterisk status
asterisk -rvvv

sip reload
sip show peers

core show applications like dial
dialplan reload
dialplan show 101@phones
dialplan show 102@phones

Register SIP Peers

Setup your SIP phone, be it soft or hard, against your registrar/domain, with the according name.

You should see some NOTICE appear. Then check,

sip show peers

You might have to,

chown asterisk:asterisk /var/lib/asterisk/astdb.sqlite3

Asterisk to Asterisk

http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html

sip show registry

TLS & SRTP

https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial

vi sip.conf

[general] … tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1

[malcolm] type=friend context=phones ;disallow=all allow=g722,ulaw,alaw secret=PASSX host=dynamic transport=tls ;dtmfmode=rfc2833

Note. cipher selection could be improved.

vi extensions.conf

[internal]
...
exten => 105,1,NoOp(malcolm is being called)
exten => 105,2,Dial(SIP/malcolm)
exten => 105,3,Hangup

Got SRTP libs at compile-time? Then,

    encryption=yes
    ;avpf=no
    ;force_avp=yes

GNU Ring

When using GNU Ring, eventually fully exit the app and re-start it, for it to appear fine as an SIP peer on the Asterisk CLI. The account name also gets fixed.

Also be very careful about the port number that you are using (advanced settings), it may conflict with the Asterisk server… Especially if you are doing TLS, 5061 is not tunable on Ring, as of today (Aug 2018). Only the non-TLS port is tunable.

Eventually start from scratch,

rm -rf ~/.config/ring/

Troubleshooting

At first run of v15 on Ubuntu Bionic, the daemon loads eventhough there is an ERROR,

WARNING[22596]: res_phoneprov.c:1230 get_defaults: Unable to find a valid server address or name.

ERROR[22596]: ari/config.c:312 process_config: No configured users for ARI

WARNING[22596]: res_hep_rtcp.c:161 load_module: res_hep is not loaded or running; declining module load

WARNING[22596]: res_hep_pjsip.c:238 load_module: res_hep is not loaded or running; declining module load

==> TODO

When reloading sip.conf, you might get this,

ERROR[32021]: chan_sip.c:4274 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

==> make sure you do not have misconfigured user-agents formaly connected.

When registering a TLS-enabled user-agent,

ERROR[17799]: iostream.c:620 ast_iostream_start_tls: Problem setting up ssl connection: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Success
ERROR[17799]: iostream.c:525 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error

==> check the client certificates and pass-phrase.

Once TLS is enabled and while setting up a call,

WARNING[18808][C-00000007]: chan_sip.c:10801 process_sdp: Rejecting secure audio stream without encryption details: audio 32638 RTP/SAVP 104 9 112 111 110 8 0 101

==> ignorecryptolifetime=yes does not help. avpf=no & force_avp=yes either. SOLUTION IS TO INSTALL SRTP LIBS AND RE-COMPILE.

Resources

Index of /pub/telephony/asterisk http://downloads.asterisk.org/pub/telephony/asterisk/

Chapter 3. Installing Asterisk http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Install.html

The Asterisk® Open Source PBX https://github.com/asterisk/asterisk/blob/master/README.md

Best Practices https://github.com/asterisk/asterisk/blob/master/README-SERIOUSLY.bestpractices.md

Asterisk Add-on Modules https://github.com/asterisk/asterisk/blob/master/README-addons.txt

SIP Configuration example for Asterisk https://raw.githubusercontent.com/asterisk/asterisk/master/configs/samples/sip.conf.sample

Dashboard https://wiki.asterisk.org/wiki/dashboard.action

Secure Real-time Transport Protocol https://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol

INSTALLING DAHDI https://support.digium.com/s/article/Installing-DAHDI

Asterisk libpri https://www.voip-info.org/asterisk-libpri/

alternatives

Gatekeeper http://fr.wikipedia.org/wiki/Gatekeeper

GNU Gatekeeper http://www.gnugk.org/