[asterisk-dev] [Code Review] Enhancements to connected line and redirecting work.
Mark Michelson
mmichelson at digium.com
Mon May 10 09:27:16 CDT 2010
On 05/09/2010 02:11 PM, Benny Amorsen wrote:
> 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
>
No, the 's' option allows the dialplan writer to modify the CID "tag" as
associated with the outgoing channel. Note that this won't actually
cause anything to be transmitted over the line. Like the description on
the review request says, a CID tag is like a channel variable, except
that it is associated with an identity (e.g. a Caller ID, Connected
Party, or Redirecting party).
The tag may be set on incoming channels through other means, such as
with func_callerid, or if the channel driver supports it, in the channel
configuration file.
Mark Michelson
More information about the asterisk-dev
mailing list