# Asterisk

Use [Asterisk](https://asterisk.org/) with **DIDWW SIP Trunking** to build
a self-managed IP PBX that routes inbound and outbound calls through Asterisk.

- Route incoming calls from DIDWW numbers to Asterisk.
- Deliver calls to internal SIP endpoints through the Asterisk dialplan.
- Use DIDWW numbers with Static Endpoint or Dynamic Registration trunks.

- Place outbound calls through DIDWW Outbound Trunks.
- Configure SIP signaling and media with Asterisk PJSIP.
- Use existing DIDWW phone numbers for inbound and outbound calling.

Note

- Choose the guide that matches the SIP channel driver used by the Asterisk
  installation. Use **Asterisk PJSIP Configuration (v20-23)** for Asterisk 21 and later. For
  Asterisk 20, use the PJSIP configuration unless the installation is explicitly
  configured with `chan_sip` and `sip.conf`.
- Asterisk deprecated `chan_sip` in version 17, stopped including it in
  default builds in version 19, and removed it entirely in version 21.



**Asterisk PJSIP Configuration (v20-23)**

Configure inbound and outbound calling for Asterisk v20-23 using `chan_pjsip` and `pjsip.conf`.

[Asterisk PJSIP Configuration (v20-23)](current-versions.html)

**Legacy Asterisk Configuration**

Use the preserved configuration for Asterisk v20 or earlier only when
the installation still uses `chan_sip` and `sip.conf`.

[Legacy Asterisk Configuration](legacy-versions.html)