[Asterisk-Dev] chan_sip / cisco sip

[Christopher L. Wade]

Is it just me, or when transferring a call on a cisco phone using sip, 
the cisco does not give *any* indication, sip message wise, of the fact 
that the call is related to an in progress call?

Looking in chan_sip.c I can see that on a successful blind transfer, the 
'dialedpeernubmer' channel variable is set, thus allowing the detection 
of who transfered the call.  But I need to know who is *transferring* 
the call :(

Is this possible?  After looking at a 'sip debug' trace of a call being 
transfered (attended/supervised) between three cisco sip phones, my 
determination is that -without looking at the sip rfc- a _truly_ key 
piece of call information is not being transmitted.  Thus making the 
answer to my question, a big bold *NO*?


Thanks,
Chris


-- 
Christopher L. Wade                     Unistar-Sparco Computers, Inc.
Senior Systems Administrator                            dba Sparco.com
Email: clwade at sparco.com                             7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400            Millington, TN 38053
Fax:   (901) 872 8482                                              USA