[Asterisk-Dev] SIP 183 early media

[Benjamin on Asterisk Mailing Lists]

On Fri, 10 Sep 2004 16:36:16 -0700, Mike Machado <mike at homelandtel.com> wrote:
> When the SIP user gets the 183 as described above, it stops providing
> ring tone to the user, so they hear dead air. If asterisk bridged the
> RTP together, the RTP coming from the gateway actually contains ringing
> tone. I know the simple answer is to let the ATA generate ring for the
> SIP user, but we have cases where its not just ring tone in the early
> media. We also do not want to start billing until the 200 comes.

Asterisk will generate CDRs for the connection with the PSTN gateway
and those are the ones you should use to determine when the PSTN
connection started, not the ones for the connection between the end
user device and Asterisk.

As far as the ringing is concerned, you can let Asterisk provide a
ringing feedback to the end user while it is trying to connect to the
PSTN gateway. In fact you could even play back a recorded message that
tells the user that Asterisk is in the process of making the
connection.

rgds
benjk

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