[asterisk-dev] Update caller information on SIP REFER

[Kevin P. Fleming]

On 08/26/2011 07:35 AM, Gunnar wrote:
> When doing an attended transfer with function keys on phones (SIP
> REFER) and not using Asterisk functions for attended transfer the
> caller information doesn't get updated. This has been discussed on
> many places, but I want to start again (cause it seems trivial to me).
>
> A calls B
> B put A on hold
> B calls C
> B connects A and C with SIP REFER
>
> Result: A and C have the caller information of B on the display.
>
> There is a function update_connectedline in chan_sip.c. It calls
> function add_rpid on a successful transfer. So A and C get's an invite
> with "Remote-Party-ID" header.
> For testing I modified add_rpid function so it adds an additional
> "P-Asserted-Identity" header after adding "Remote-Party-ID". After
> this change I see on phone A the caller information of C and on phone
> C the caller information of A! Tested with snom and Aastra phone.
> Trivial change, but a bad hack I think. Every time a "Remote-Party-ID"
> is added I have the "P-Asserted-Identity". Maybe not the best
> behaviour in all situations ...
> I'm not a big programmer in Asterisk, but my point of view is this:
> Change function update_connectedline and set a flag. In function
> add_rpid check this flag. If set add additional "P-Asserted-Identity"
> header.
>
> Anyone? Suggestions?

Did you read the sample configuration file? If you set "sendrpid=pai", 
the code will send P-Asserted-Identity instead of Remote-Party-ID.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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