ITC-2300 VoIP Lab Sample pjsip.conf File

The following shows an example of the things you may need to change in your pjsip.conf file.

Example Minimal pjsip.conf File Changes

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

[2903] ; The value inside the [] will be the username on the device
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 name used by the endpoint
username=debra ; This is the authentication password used by the endpoint

Voicemail pjsip.conf File Changes

You can add the ability to enable the Message Waiting Indicator (MWI) lamp on SIP phones when there is a message in a mailbox. By default Asterisk will send SIP NOTIFY messages when a voicemail is left. Some phones need to subscribe to the MWI information. In that case you need to modify the AoRs section for the phone and add a mailboxes= section.

For Example:

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

DPMA pjsip.conf File Changes

In order to use the DPMA with the chan_pjsip module a default_outbound_endpoint needs to be enabled in pjsip.conf. This is accomplised by configuring a basic endpoint to use and then setting that endpoint as the default.

For Example:

[global]
type=global
default_outbound_endpoint=dpma_endpoint

[dpma_endpoint]
type=endpoint

You will also need to create a SIP line for the phone to use.

For Example:

[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>
disallow = all
allow = alaw

ISDN Router <-> SIP pjsip.conf File Changes

[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_external
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

IP Telephony Service Provider (ITSP) PSTN pjsip.conf File Changes

You will need to modify the username for registration, the authentication username and password, and the endpoint user to be your Pod so you receive calls to the correct numbers!

[simpletrans]
; We need to make some changes here because we are behind NAT so that when we register to the ITSP they get our public IP (which is being forwarded to Asterisk by our router) instead of our internal IP address
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.10.0/24 ; Set the IP address(es) for the local network, these devices will use the actual IP address of the Asterisk server
; Change these IP addresses to point to the ESXi-1 address for your pod (and which your router is forwarding the ports through to Asterisk)
external_media_address=172.17.144.XX ; This IP addresses will be used for handling audio (media) with people outside our local network (like the ITSP)
external_signaling_address=172.17.144.XX; This IP addresses will be used for handling audio (media) with people outside our local network (like the ITSP)

[sipPSTN]
type=registration
transport=simpletrans
outbound_auth=sipPSTN
server_uri=sip:172.17.139.25 ; The IP address of the ITSP where we should register our IP address so they know where to send calls
client_uri=sip:pod8@172.17.139.25 ; 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=pod8-password
username=pod8

[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 behind NAT and only have ports forwarded to the Asterisk server
from_user=pod8 ; 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:172.17.139.25:5060 ; All calls to the ITSP will go to this IP address and port

[sipPSTN]
type=identify
endpoint=sippstn
match=172.17.139.25 ; All calls from the ITSP will come from this IP address

Reloading After Configuration Changes

If you make changes to the pjsip.conf file and need to get Asterisk to re-load the configuration you can run the pjsip reload command from the Asterisk CLI to re-read the file.

Additional Resources

• PJSIP Config Sections and Relationships
• Core PJSIP Configuration Options
• SIP Configuration for DPMA Phones