Hi List,
I am research in Brazil University and in our system we have several
User Agents in SIP.
We have the openser working as Sip Proxy and Asterisk or Cisco Routers
as Gateway with PSTN or PBX.
We have too Hardphone Cisco IP Phone and Polycom IP Pphone and
Softphone X-Lite, Eyebeam and Ekiga.
Just now our system is working to all users, but I found one problem.
When the user, work with BlindTransfer* the call is transfered no problem.
But when the user (no matter if is Hardphone or Softphone) try
transfer** the call, it don't work.
In my OpenSER, when I receive one REFER without LOOSE_ROUTE I just end
the request.
While if this request have LOOSE_ROUTE I send this request to route
defined in dialog.
I would like if you can I help me, if I can insert one logical in
OpenSER to do it work.
is it possible?
Very thanks for everybody,
Thiago!
(!)
* BLINDTRANSFER: UA1 call to UA2 and when the call is established, UA1
for example send one request REFER to UA2 to this user call UA3. Then
UA2 hangup call with UA1, and make one call to UA3.
** TRANSFER: UA1 call to UA2 and when the call is established, UA1 for
example put this call in hold on, and make one call to UA3. When this
last call is established, UA1 send one request REFER to UA2 to get the
call with UA3. Then UA2 will talk with UA3, and UA1 is hangup its
calls.
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THIAGO MALUF RESENDE
Consultor Voip e Programador WEB (Voip Developer and Web Developer)
Tel: +55 21 86042100
e-mail: malufrj(a)gmail.com