[Asterisk-Dev] [Asterisk-Users] Attended transfer with original CID info. NEW FEATURE?

[Jared Mauch]

On Fri, Jul 22, 2005 at 09:38:50AM +0200, Kib Eki wrote:
> Kevin,
> 
> do you know of anyone who tried to do this for a special phone?

	This should work with the 7970 with chan_sccp.

sip-server*CLI> show application SetCalledParty
  -= Info about application 'SetCalledParty' =- 

[Synopsis]
Sets the name of the called party

[Description]
  SetCalledParty("Name" <ext>) sets the name and number of the called party for use with chan_sccp


	I think that all you might need to do after this is implement
the SIP caller-id updating...

	- jared

> Kevin P. Fleming wrote:
> >Kib Eki wrote:
> >
> >>is it possible to do an attended transfer so that the original CID 
> >>info will
> >>stay for that call.
> >
> >
> >It's possible (nearly anything is possible), but completely 
> >unimplemented. Keep in mind that it will only be possible for endpoints 
> >(phones) whose CID display can be changed _after_ the call is already 
> >connected.
> >
> >>Is this a new feature which needs a bounty?
> >
> >
> >That would be a wise idea, if there is anyone out there in the community 
> >besides me who wants to work on it. I can't accept a bounty, for obvious 
> >reasons :-)
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-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.