Asterisk SIP channels

SIP Channel Module The SIP Channel Module enables Asterisk to communicate via VOIP with SIP telephones and exchanges. Asterisk is able to act as

a SIP client: This means that Asterisk registers as a client to another SIP server and receives and places calls to this server. Incoming calls are routed to an Asterisk extension.
a SIP server: Asterisk can be configured so that SIP clients (phones, software clients) register to the Asterisk server and set up SIP sessions with the server, i.e. calls and answers incoming calls. This said, Asterisk is not a full-feature SIP server like SIP express router or OpenSER. If you are going to have thousands of SIP phones, you should use SER or OpenSER and forward calls to Asterisk for voicemail or PSTN access.
a SIP gateway: Asterisk acts as a Media gateway between SIP, IAX, MGCP, H.323 and PSTN connections. As an example, an Asterisk server can be connected to ISDN to give your SIP clients connectivity to the switched telephony network.
Why is Asterisk not a SIP proxy?

Notes

Asterisk supports ENUM — see Asterisk config enum.conf.
Asterisk support DNS SRV records, pointers to SIP proxy servers for Internet domains. See Asterisk SIP srvlookup
When and why is Asterisk a media gateway for SIP calls? - see Asterisk SIP media path
With a bit of tweaking, Asterisk supports SIP URL dialing.
Asterisk supports SIP over UDP, but not over TCP.
You can store your SIP devices in a database for dynamic configuration, see Asterisk SIP Mysql Peers

Configuring SIP Channels Configuration of SIP Channels is done by modifying the sip.conf file. See:

SIP Channel Configuration

Using the Dial Command with SIP Channels Recall that the format of the Dial command is like this:

Dial(type/identifier,timeout,options,URL)

For SIP channels, the type is always SIP. The timeout, options and URL parts are explained on the Dial page. ...