[SIPForum-discussion] Call transfer

[Abinash Sarangi]

Here you should get a 200 OK for re-invite ..no 180 Ringing or 183 Session Progress.
Would it have been a transfer from audio to video call or some kind of media after the established session[usually IVR/Announcement], then you should expect 18x message.
Why you need to send a Supported: 100rel in re-invite..is it any specific scenario.
If there is a non-failure and reliable response from the UAS then no need to handle PRACK..
 
 
Thanks & Regards -Abinash > Date: Sat, 17 Jan 2009 13:51:25 +0530> From: rout.shibashankar at gmail.com> To: nabamserbang at yahoo.com> CC: discussion at sipforum.org> Subject: Re: [SIPForum-discussion] Call transfer> > Hello All,> > I am getting stucked at this scenario. Can anybody help me on this ?> > My question : Suppose i am sending re-Invite without SDP with> supported header 100rel to UAS. Then shall the UAS send 180/183> Ringing with SDP offer for enabling the PRACK?> > Please clarify my doubt. My doubt is after a established session ,> shall i get 180 ringing for Re-invite. ?> > > Any suggestion would be vital for me ?> > > Invite---->> 180------>> 200----->> Ack----->> ====> Re-Invite---->> 180----------->> Prack----->> > > > Does this call flow right ??> > > Regards,> S S Rout> > > On 1/16/09, nabam serbang <nabamserbang at yahoo.com> wrote:> >> > Seems you are sending some sort of line appearance selection feature in INV> > to C at step 5 or there is already one call with C.> >> > Whatever be the case, it doesn't really matter on which line call comes so> > long call work ( I suggest you to see if any line selection feature is being> > sent on step 5). And Your call flow MUST execute steps 6, 7 and 8 as well> > for complete transfer.> >> > Thanks> > ~Nabam> >> >> >> >> > ________________________________> > From: Naga Yerramsetti <naga95 at gmail.com>> > To: nabam serbang <nabamserbang at yahoo.com>> > Sent: Friday, January 16, 2009 12:44:05 AM> > Subject: Re: [SIPForum-discussion] Call transfer> >> > I am having problem at step 5. C is getting the invite on second line> > instead of getting connected on the first line.> >> >> > On Thu, Jan 15, 2009 at 1:48 AM, nabam serbang <nabamserbang at yahoo.com>> > wrote:> >> >> >> >> > 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> >> >> >> _______________________________________________> This is the SIP Forum discussion mailing list> TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion> Post to the list at discussion at sipforum.org
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