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It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.
When reading the instructions below be aware which are for sip.conf and which are for pjsip.conf
=pjsip.conf (PJSIP)=
==PJSIP: Trunk registration==
Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.
In pjsip.conf:
[reg_442082881111]
type = registration
retry_interval = 20
fatal_retry_interval = 20
forbidden_retry_interval = 20
max_retries = 9999
auth_rejection_permanent = no
contact_user = maininbound
expiration = 120
outbound_auth = auth_reg_442082881111
client_uri = sip:+442082881111@voiceless.aa.net.uk
server_uri = sip:voiceless.aa.net.uk
[auth_reg_442082881111]
type = auth
password = BusinessPaidGrewCome
username = +442082881111
[aaisptrunk]
type = aor
contact = sip:+442082881111@voiceless.aa.net.uk
qualify_frequency=20
[aaisptrunk]
type = identify
endpoint = aaisptrunk
match = voiceless.aa.net.uk
[aaisptrunk]
type = endpoint
context = maininbound
dtmf_mode = rfc4733
disallow = all
allow = alaw
allow = ulaw
direct_media = no
rtp_symmetric = yes
aors = aaisptrunk
outbound_auth=auth_reg_442082881111
Calls come into the context "maininbound" in extensions.conf - in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.
[maininbound]
exten = maininbound,1,Dial(PJSIP/222&PJSIP/205,20)
exten = maininbound,n,Voicemail(222@default,us)
In extensions.conf you can dial out via the trunk with:
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
exten => s-BUSY,1,Playtones(busy)
exten => s-CONGESTION,1,Playtones(congestion)
exten => s-CHANUNAVAIL,1,Playtones(unobtainable)
exten => s-NOANSWER,1,Playtones(congestion)
==PJSIP: Trunk without registration==
Use the above example but do not include the top section for "[reg_442082881111]".
Then set the AAISP control panel to point to your server by hostname or IP address:<br />
[[File:Asterisk pjsip noregistration.png|border]]
==PJSIP: Keep-Alive / Anti-Idle==
If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.
Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds.
An example of the aor section follows:
[aaisptrunk]
type = aor
contact = sip:+442082881111@voiceless.aa.net.uk
qualify_frequency=20
==PJSIP: IPv6==
By default PJSIP does not listen on IPv6.<br />
At the top of pjsip.conf you will need to add another transport to go along with your IPv4 transport (usually a section with bind set to 0.0.0.0 or your IP address):
[transport-udp6]
type=transport
protocol=udp
bind=[::]
You may need to force the endpoint to use this transport (a small section of the endpoint section above but with the "transport = " line inserted):
[aaisptrunk]
type = endpoint
transport = transport-udp6
==Status and Commands==
A good command within the asterisk software is the show registration command:
asterisk*CLI> pjsip show registrations
<Registration/ServerURI..............................> <Auth..........> <Status.......>
==========================================================================================
reg_442082881111/sip:voiceless.aa.net.uk auth_reg_442082881111 Registered
Objects found: 1
In this example it shows that the Asterisk server is successfully registered with the Andrews & Arnold SIP server.
=sip.conf (SIP)=
exten => _X.,1,Dial(SIP/snom)
</syntaxhighlight>
=Firewall & Security=
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