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also the type of service you would get if you were simply registering a SIP phone
directly to a service.
IP-based authentication
No password; no registration. This is more common with carriers that provide
bulk trunking services to larger enterprises and resellers. (Typically these will also
come with some sort of minimum commitment in terms of volume.) You will be
expected to have solid SIP and networking skills.
These are not hard-and-fast rules, but they are the most common in our experience.
So there are two ways we might configure an ITSP
pjsip_wizard.conf file.
in the /etc/asterisk/
First, if the carrier uses an IP address–based authentication, they will expect you to
send your traffic from a static IP address (and should your address change, you will
need to inform them so they can reconfigure their equipment). Your pjsip_wiz‐
ard.conf file could then look something like this:
; ITSP uses IP address-based authentication
[itsp-no-auth]
type=wizard
remote_hosts=itsp.example.com
endpoint/context=pstn-in
endpoint/allow = !all,ulaw,g722
sends_registrations=no
accepts_registrations=no
15 Remember that registration is simply a mechanism whereby a SIP endpoint tells a registrar server where it is
located. This is useful if your IP address changes, as might be the case on a consumer or small-business type
of internet connection (such as DSL or cable).
VoIP
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sends_auth=no
accepts_auth=no
Alternatively, if your IP address changes frequently (or your carrier requires this
method), you can have your system register to the carrier (which will require you to
send authentication credentials to prove it’s really you). Your calls will typically also
be required to authenticate:
[itsp-with-auth]
type=wizard
remote_hosts=itsp.example.com
endpoint/context=pstn-in
endpoint/allow = !all,ulaw,g722
sends_registration=yes
accepts_registrations=no
sends_auth=yes
accepts_auth=no
outbound_auth/username=itsp_provided_username
outbound_auth/password= itsp_provided_password
Note that the names [itsp-no-auth] and [itsp-with-auth] have no built-in mean‐
ing to Asterisk. They become the PJSIP channel names to which you send your calls.
Configure trunks for termination. The PJSIP wizard has created the channel definitions
we require for our carrier. To send a call, we only need to make a minor change to the
[globals] section of our extensions.conf file, as follows:
[globals]
UserA_DeskPhone=PJSIP/0000f30A0A01
UserA_SoftPhone=PJSIP/SOFTPHONE_A
UserB_DeskPhone=PJSIP/0000f30B0B02
UserB_SoftPhone=PJSIP/SOFTPHONE_B
TOLL=PJSIP/itsp-no-auth
LOCAL=${TOLL}
;OR
;TOLL=PJSIP/itsp-with-auth
;LOCAL=${TOLL}
Configuring trunks for origination. For your incoming calls, you’ll need a context in
your /etc/asterisk/extensions.conf file that matches the context specified for the ITSP
channel. Let’s assume we have two NANP DIDs, 4169671111 and 4167363636, and
place the required code above the [sets] context:
TOLL=PJSIP/itsp-no-auth
LOCAL=${TOLL}
;OR
;TOLL=PJSIP/itsp-with-auth
;LOCAL=${TOLL}
[pstn-in]
exten => 4169671111,1,Dial(sets,100,1)
exten => 4167363636,1,Dial(sets,101,1)
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Chapter 7: Outside Connectivity