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  ; 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 < >)  ; 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.
 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.
[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.
[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_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
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:email@example.com ; 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.