Lab 5 Using PJSIP

These labs are part of the training Complete Asterisk Training at www.udemy.com, attend using our discounted coupon

Lab 5 - SIP using the new channel driver PJSIP.

Note: When testing this lab I saw sometimes XLITE loosing the registration. So check if phones registration is ok before inbound tests using pjsip show aors

Step 1 - Disable chan_sip and load chan_pjsip

Edit /etc/asterisk/modules.conf and change the following lines.

;noload => chan_pjsip.so
noload =>chan_sip.so

Go to the Linux command line and restart Asterisk

#/etc/init.d/asterisk restart

Step 2 - Save the original pjsip.conf

Step 3 - Configure the new pjsip.conf

[transport-udp-main]
type=transport
protocol=udp
bind=0.0.0.0:5060
[transport-udp-alternative]
type=transport
protocol=udp
bind=0.0.0.0:5080

[xlite]
type=endpoint
transport=transport-udp-main
context=from-internal
disallow=all
allow=ulaw
aors=xlite
auth=xlite
[xlite]
type=auth
auth_type=userpass
username=xlite
password=#supersecret#
[xlite]
type=aor
max_contacts=2

[zoiper]
type=endpoint
transport=transport-udp-main
context=from-internal
disallow=all
allow=ulaw
aors=zoiper
auth=zoiper
[zoiper]
type=auth
auth_type=userpass
username=zoiper
password=#supersecret# [zoiper]
type=aor
max_contacts=2

[blink]
type=endpoint
transport=transport-udp-main
context=from-internal
disallow=all
allow=ulaw
aors=blink
auth=blink [blink]
type=auth
auth_type=userpass
username=blink
password=#supersecret#
[blink]
type=aor
max_contacts=2

Step 7 - Change the naming of the SIP channels in extensions.conf

[globals]
OPERATOR=PJSIP/xlite

exten=>6000,1,gosub(PJSIP/zoiper,${EXTEN})
exten=>6001,1,gosub(PJSIP/xlite,${EXTEN})
exten=>6002,1,gosub(PJSIP/blink,${EXTEN})
exten=>6003,1,gosub(PJSIP/bria,${EXTEN})

exten=>1,1,dial(PJSIP/zoiper)
exten=>2,1,dial(PJSIP/xlite)
exten=>3,1,dial(PJSIP/bria)
exten=>6000,1,Dial(PJSIP/zoiper)
exten=>6001,1,Dial(PJSIP/xlite)

Step 8 - Test calls between extensions, please reload all phones. Pay attention to disable stun on all phones

Note: Some times, only one softphone get access to the multimedia, so try from all the phones, sometimes only one of them will get audio.

Step 9 - Create the siptrunk

Edit the file /etc/asterisk/pjsip.conf

[siptrunk]
type=endpoint
transport=transport-udp-main
context=from-siptrunk
direct_media=no
disallow=all
allow=ulaw
outbound_auth=siptrunk
aors=siptrunk

[siptrunk]
type=aor
contact=sip:sip.flagonc.com:5600

[siptrunk]
type=auth
auth_type=userpass
username=1020
password=supersecret

[siptrunk]
type=registration
outbound_auth=siptrunk
server_uri=sip:[email protected]:5600
client_uri=sip:[email protected]
contact_user=9999

[siptrunk]
type=identify
endpoint=siptrunk
match=sip.flagonc.com

Step 10 - Change the naming of the SIP channels in extensions.conf

exten=>_9.,1,dial(PJSIP/${EXTEN:1}@siptrunk,20)

Step 11 - Test the outgoing calls

Dial 9130523456789

Step 12 - Test the incoming calls

Go to the page sip.flagonc.com and use the click to call to generate a call to your registered account (in the lab case 1020). Check if monkeys called your xlite phone as they can't choose an option in your IVR :)