PBX interface function -build SIP pvt structure

Note:: SIP calls initiated by the PBX arrive here

The SIP dial string is what is used in the dial plan, extensions.conf.

Currently:

DIAL(SIP/user)
DIAL(SIP/user)
DIAL(SIP/domain/exten)

In SIP3:

DIAL(SIP/sip:username!destination)
- Any SIP uri

DIAL(SIP/object/extension!<destination>)
- object can be a phone or a trunk
- extension can be any sip uri
- if domain is not used, the domain/host of the object is used as the request URI
- Like DIAL(sip/trunk5/oej.net)
  - We assume a sip: uri unless a scheme is set

This means that old dial plans will not be compatible with the new channel.

The dialstring can be divided into parts

object name
uri

The To: header and the request URI

It's not easy to determine the contents of the to: header. Will it be the same as the extension dialled into the PBX?

The To: header will contain the extension dialed. The DNID given in the destination option will be used in the To: header, with local domain or domain included in DNID

dial(SIP/object!user.com)

will dial the object and use To: user@example.com in the To:-header

dial(SIP/object!user)

Will have To: user

dial(/SIP/sip:user!12345)

Will use "user@domain" as the Request URI and set 12345 as the destination of the call in the To: header

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Asterisk developer's documentation :: Codename Pineapple

chan_sip3 :: The SIP dial string