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

[Andrew Latham]

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


~
Andrew "lathama" Latham
lathama at gmail.com

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On Sun, May 9, 2010 at 1:34 PM, Benny Amorsen <benny+usenet at amorsen.dk> wrote:
> "Mark Michelson" <mmichelson at digium.com> writes:
>
>> Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
>> caller ID tag to be placed on the outgoing channel.
>
> I am probably just dense, but what does this offer which just setting
> CALLERID() cannot do?
>
> We are in the strange position of often having to set CALLERID(num)
> differently for two dial targets when doing Dial(a&b), so anything
> which is per-dial-statement does not really work for us.
>
>
> /Benny
>
>
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