Thought about converting across to PJSIP? here are some helpful hints and configuration examples to connect your vanilla Asterisk to our environment.
The main part of the conversion is the population of the pjsip.conf file. There will also need to be changes made to your extensions.conf file to dial out using the PJSIP channel’s.
To start with you will need to get your system to register and set up a contact/AOR for Simtex.
For this to work we need to set up the following:
- A Registration type
- An Auth type
- An AOR type
To be able to use this registration you will need an endpoint associated and also and Identify type.
| 1. | Setting up an Registration Type:
[Simtex]
type = registration
server_uri=sip:[email protected]
client_uri=sip:[email protected]
contact_user=7XXXXXXX
outbound_auth=Simtex_Auth
auth_rejection_permanent = no
retry_interval = 30
forbidden_retry_interval = 290
max_retries = 20
|
| 2. | An Auth type
[Simtex_Auth]
type=auth
username=7XXXXXXX
password=XXXXXXXX
|
| 3. | An AOR type [Simtex_Aor] type=aor contact=sip:sip.simtex.com.au qualify_timeout=4.0 qualify_frequency=50 |
From here we need to set up a transport for an endpoint to use after that an endpoint and a Identify type can be set up.
| 1. | Transport:
[Transport-UDP]
type=transport
protocol=udp
bind=0.0.0.0
Note: This is a very simple Transport and may need to be change based on your own setup.
|
| 2. | Endpoint type:
[Simtex_Endpoint]type=endpoint |
| 3. | Identify type:
[Simtex_Identify] |
Now that we have that setup an extension will need to be configured to be able to make and receive calls form the Simtex endpoint.
The following is a basic extension and will need to be changed based on your particular setup: