[asterisk-dev] [Code Review] Enhancements to connected line and redirecting work.

Benny Amorsen benny+usenet at amorsen.dk
Sun May 9 14:11:10 CDT 2010


Andrew Latham <lathama at gmail.com> writes:

> Summary: The SIP phone can display the ID of the called party after
> connection.  You can get the Caller information updated for each
> party.

So the s option changes what is sent on the "incoming" channel, not the
outgoing one?


/Benny




More information about the asterisk-dev mailing list