[SIPForum-discussion] Call transfer

[naga95]

I am having  problem at step 5.  C is getting the invite on second line
instead of getting connected on  the first line.
 
  _____  

From: nabam serbang [mailto:nabamserbang at yahoo.com] 
Sent: Thursday, January 15, 2009 1:49 AM
To: Naga Yerramsetti; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Call transfer
 



     A        B2BUA          B               C

               dlg1 (A-B call)
 1) <--------------------->
 
      Hold dlg A-B (When C preses tranfer, it generates soft hold to A)

 2) <----------------------

                    B --> C ( B establishes call with C) dlg2   
 3)            <------------
                 -------------------------->                   

             REFER (to C from B as B has initiated transfer to C)
              Refer-to:c@<c-address>?Replaces=dlg2
 4)<---------------------     

     INVITE sip:c@<c-address>
     Replaces:dlg2   
 5) ---------->-------------------------->
         200 OK     
    <---------------------------------------
 
      NOTIFY (for transfer) 
 6) --------->--------------->

         BYE (to B) dlg1
 7)  --------->----------->

                    BYE (to C) dlg2 
 8)              ------------------------->     



 
Steps 4 and 5 in above call flow should solve your problem. dlgx is one
complete call setup steps i.e. from INVITE to ACK.


 Thanks and regards
 Nabam, Avaya
 Home page: http://sites.google.com/site/nabamserbang/
  _____  

From: Naga Yerramsetti <naga95 at gmail.com>
To: discussion at sipforum.org
Sent: Wednesday, January 14, 2009 9:42:34 PM
Subject: [SIPForum-discussion] Call transfer
Hi,
I am a newbie in SIP. We are using web logic sip server as B2BUA.
 
I am having problem implementing call transfer. Call is established between
A and B
B is trying to transfer A to C. Call is established between B and C.
Call leg to A and Call leg to B exists in one application session and a call
leg to B and call leg to C exists in another  sip application session.
Now what the B2bUA (app running in weblogic sip server) should do to
establish call between A and C
 
Thank you. 
Naga
 
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