When configuring a SIP trunk in Asterisk, it's important to choose the right channel module. chan_pjsip is recommended over the older chan_sip due to its better performance and support for modern SIP features. This guide walks you through setting up a SIP trunk with Internetport using chan_pjsip.

Difference between chan_sip and chan_pjsip

To understand why chan_pjsip is the recommended choice, it's important to know the differences. chan_sip is the older channel module with limited support for modern SIP features and is less flexible than chan_pjsip. In contrast, chan_pjsip is built on the PJPROJECT library, which offers better performance, more advanced features, and support for SIP advanced functions like WebRTC.

Configure /etc/asterisk/pjsip.conf

Step 1: Transport
Start by defining the transport in pjsip.conf. This specifies how communication occurs between Asterisk and the SIP server.
```
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
```

Step 2: Endpoint

The next step is to create an endpoint representing the SIP trunk with Internetport.

[internetport]
type=endpoint
context=default
disallow=all
allow=g711a,g711u,g722,g729
transport=transport-udp
auth=internetport_auth
aors=internetport_aor

Step 3: Auth
To authenticate with Internetport's SIP server, you need to specify the username and password.
```
[internetport_auth]
type=auth
auth_type=userpass
username=ditt_anvandarnamn	password=ditt_losenord
```
Step 4: AOR
AOR (Address of Record) is used to manage SIP addresses and registration.
```
[internetport_aor]
type=aor
contact=sip:sip.internetport.se:5060
```

Configure Registration with sip.internetport.se

To enable automatic registration with Internetport's SIP server, add the following to pjsip.conf.

[internetport]
... ; previous configuration
outbound_auth=internetport_auth
registration=sip:sip.internetport.se:5060

Dialplan in extensions.conf for Incoming and Outgoing Calls

To handle incoming and outgoing calls, configure the dialplan in extensions.conf.

Incoming Calls

[default]
extension => 1000,1,NoOp(Incoming call)
same => n,Dial(PJSIP/internetport/${EXTEN})

Outgoing Calls

[default]
extension => _X.,1,NoOp(Outgoing call)
same => n,Dial(PJSIP/internetport/${EXTEN})

Reload Asterisk Modules

Once the configurations are complete, reload Asterisk's modules to apply the changes.

asterisk -rx 'core reload'

Verify Registration

To verify that the SIP trunk is correctly registered, use the following command in the Asterisk CLI:

pjsip show registrations

Troubleshooting

Check logs for any error messages.
Use pjsip set logger on to enable detailed logging and identify issues.

For more help, visit Internetport's support pages.