Re: [OpenSIPS-Users] B2BUA help

2010-02-09 Thread marie.gremillot
Hello every body,

I have installed opensips and b2bua and I often have these errors (followed by 
the crash of the server) :

segfault at 646e657b ip 080de8dc sp bfbcceb0 error 6 in opensips[8048000+10f000]
CRITICAL:core:receive_fd: EOF on 15


It appears at different moments : when I connect, when I call, when the server 
is up too many time...





 Message du 08/02/10 18:43
 De : Anca Vamanu
 A : OpenSIPS users mailling list
 Copie à :
 Objet : Re: [OpenSIPS-Users] B2BUA help


 opensipsl...@encambio.com wrote:
  Hello Anca,
 
  An lun., févr 08, 2010, Anca VAMANU schrieb:
 
  The scenario that you want - authorizing the user at the media
  server in a bridging scenario will not work. You can not call
  b2b_init twice for the same dialog. The B2B implementation does not
  include the authorization requirements. And in fact I believe that
  you are complicating things by wanting the user to authorize to the
  media server. The normal approach will be for the user to authorize
  to the proxy and only after it has been authorize to call b2b_init
  function. And for the media server not to ask for authorization
  because the proxy should be in a trusted source list and the
  requests coming from it should not be challenged.
 
 
  Okay now it's clear, thanks for the help. But the problem is not
  solved, because there are situations in which the INVITE from your
  B2B logic should be challenged. In my case, the media server is
  serving voicemail as well. It must challenge incoming INVITEs
  because sometimes the 'From' URI describes a voicemail user that
  will be authorized to listen to their messages by way of the
  standard WWW-Authenticate.
 
  At the same time, I'm sure there's no problem in passing the
  'Authorization' header if I make a new B2B module or simply patch
  the existing one. Is it true that calling b2b_init a second time
  will be no problem, because as b2b_logic sees the 401
  WWW-AUthenticate it will close the dialog started by the INVITE?
 
  Can you please advise me whether the needed change to pass the
  'Authorization' header is in b2b_entities or b2b_logic? Do you
  see any problem in doing this?
 
  I have some other ideas of implementing B2B modules ontop of
  b2b_entities, so this would be a good way to become introduced
  to the code.
 
  Regards,
  Brian
 
 Yes, the authorization header should pass no matter what. I will try
 this myself and check if there is a bug.

 Regards,

 --
 Anca Vamanu
 www.voice-system.ro


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[OpenSIPS-Users] issue with my XML script, b2bua

2010-02-05 Thread marie.gremillot
hello,
I've written a XML script that should do that :

opensips                  b2bua                                 
   AS
                -(forward 
INVITE)-
    -(200 OK) 

    (forward INVITE) -

Here is the script :

?xml version=1.0?

 
 

    
    server1
    
    
    client1
    message
    
    0
    
    
 
 1
 
 
 
 
 
 
 1
 
 client
 client1
 
 
 
 
 200
 OK
 
 
 

 
 client
 server1
 
 
 client1
 
 server1
 
 
 
 
 

    
    
    





When I do the call, I can see that:
- the invite is forwarded to the AS
- the AS answer a 200 OK
- but the phone call, but the callee doesn't receive the call

thanks !


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[OpenSIPS-Users] b2b_key match found

2010-02-05 Thread marie.gremillot
I have this error when I make a call via a b2b :

ERROR:b2b_logic:b2b_logic_notify: No b2b_key match found

I don't find anything on it on the internet ...

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[OpenSIPS-Users] B2Bua : send 2 INVITE without caring about the 200 OK

2010-01-29 Thread marie.gremillot
hello,
I have a question concerning the B2Bua : I'd like to send 2 INVITE without 200 
OK. Indeed I'd like to send the second INVITE even if I haven't received the 
200 OK of the first one.
But I find that the B2Bua is too much sequetial
Is there a solution ?

thank you


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[OpenSIPS-Users] Parameters in b2bua

2010-01-26 Thread marie.gremillot
hello, I'm using B2bua and I'd like to put instead of the parameters something 
like rU (the callee) in order to get back the first part of his SIP address 
(for instance bob when his SIP adress is sip:b...@ims.com) . But I don't know 
how to do it (when I tried to put rU, I had many errors. So I add quote, but 
it uses rU and not what rU means )
I don't know if I'm clear ...


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