ITC-2300 VoIP Lab Sample pjsip.conf File

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

[2903] ; The value inside the [] will be the SIP line user name on the endpoint
type=endpoint
context=default
disallow=all
allow=ulaw
transport=simpletrans
auth=debra-auth ; This will be the name for the authentication section of the configuration found below
aors=2903 ; This will be the name for the AoRs registration section of the configuration found below it must match the username in the [ ] above
trust_id_outbound=yes
callerid=Debra Donaldson <2903> ; This is the caller ID name and extension (in the < >)

[2903] ; This is the AoRs registration entry. The value in the [ ] must match the endpoint aors line and username on the device
type=aor
max_contacts=1

[debra-auth] ; This is the authentication entry. The value in the [ ] must match the endpoint auth line
type=auth
auth_type=userpass
password=debrapass ; This is the authentication password used by the endpoint for sip registration
username=debra ; This is the authentication name used by the endpoint for sip registration

[2903]
type=aor
max_contacts=1
mailboxes=2903@default

[global]
type=global
default_outbound_endpoint=dpma_endpoint

[dpma_endpoint]
type=endpoint

[echo2904]
type = aor
max_contacts = 1
mailboxes = 2904@default

[echo2904]
type = auth
auth_type = userpass
password = 2904
username = 2904

[echo2904]
type = endpoint
aors=echo2904
auth=echo2904
callerid=Echo Early
context = default
direct_media = yes
trust_id_inbound = yes
trust_id_outbound=yes
send_pai = yes
transport=simpletrans
callerid=Echo Early <2904> ; This is the caller ID name and extension (in the < >)
disallow = all
allow = alaw

[ISDNrouter]
type=endpoint
; You want to choose a context that any calls coming in from the gateway will go to and be careful to limit where these calls can go so that people can't "loop through" your Asterisk system and run up your phone bill or otherwise manipulate the system
context=from_PSTN
disallow=all
allow=ulaw
aors=ISDNrouter

[ISDNrouter]
; This will send SIP traffic to our router’s IP address if it’s sent to the ISDNrouter SIP device
type=aor
contact=sip:192.168.10.1:5060

[ISDNrouter]
; This will match the SIP traffic incoming from the ISDN line attached to our router
type=identify
endpoint=ISDNrouter
match=192.168.10.1

[sipPSTN]
type=registration
transport=simpletrans
outbound_auth=sipPSTN
server_uri=sip:172.17.219.2 ; The IP address of the ITSP where we should register our IP address so they know where to send calls
; Update the username in the following line for your pod!
client_uri=sip:podX@172.17.219.2 ; The username we need to provide to register with the ITSP as well as the IP address of the ITSP registration server
endpoint=sipPSTN
line=yes

[sipPSTN]
type=auth
auth_type=userpass
; Be sure to modify the username and password as appropriate for your pod
password=podX-password
username=podX

[sipPSTN]
type=endpoint
transport=simpletrans
context=from_external ; What context calls from the ITSP should end up in
disallow=all
allow=ulaw
outbound_auth=sipPSTN
aors=sipPSTN
direct_media=no ; This line tells Asterisk to not just handle the call setup information but to pass the audio through Asterisk as well and is required because we are using Asterisk to connect between the public network (ens192) and our pod network (ens224) where our phones are
from_user=podX ; The username to use in the SIP messages for calls out to the PSTN be sure to set correctly for your pod!

[sipPSTN]
type=aor
contact=sip:172.17.219.2:5060 ; All calls to the ITSP will go to this IP address and port

[sipPSTN]
type=identify
endpoint=sippstn
match=172.17.219.2 ; All calls from the ITSP will come from this IP address