[asterisk-dev] limit sip call transfer after refer message

[Matthew Jordan]

On 10/04/2012 08:58 AM, nicolo' wrote:
> a question, on asterisk you can limit a consultation call transfer on
> bridge channels?
> I've tried using a particular context set with the variable
> TRANSFER_CONTEXT but this only works with blind transfers ..
> 
> On asterisk 1.8.16,chan_sip.c function get_refer_info, I forced to use
> TRANSFER_CONTEXT but with no results..
> 
> other ideas?

Please ask this question on asterisk-users.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org