[OpenSIPS-Users] [NEW] exchanging info between dialogs

[erik pepermans]

Hi,

I have the following scenario :


A1 asterisk server initiates a call to A2 asterisk server thru opensips;
This A2 calls A and A lands on A1 asking A to dial a number. A then
initiates a new call to A2 asterisk server thru opensips which calls B. A
talks to B.

The issue is that when B hangs up the 'BYE' message is not sent to A1, but
twice to A2. The session on A1 hangs forever.

Does the below fix this ?

Brgds
Erik 

-----Oorspronkelijk bericht-----
Van: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Namens Bogdan-Andrei Iancu
Verzonden: woensdag 28 april 2010 17:47
Aan: users at lists.opensips.org; devel at lists.opensips.org;
news at lists.opensips.org
Onderwerp: [OpenSIPS-Users] [NEW] exchanging info between dialogs


Hi,

just added to the dialog module a new function that allow you to 
exchange data between dialogs - mainly to extract data from a different 
ongoing dialog.

Such functionality is vital in complex scenarios (PBX related) like 
attended call transfer - in such cases you may want to route a new call 
based on information of existing dialogs.

Real case example:

    OpenSIPS is doing dispatching over a set of Asterisk boxes (which 
act as SIP servers).
    A calls B and the call is established (by dispatching from OpenSIPS) 
via A1 Asterisk server
    A wants to transfer B to a new party C, so A makes a new call to C 
-> this call must end on A1 also, without going via dispatcher in openSIPS.
    So, when A calls C, OpenSIPS will check if A has an already existing 
call and if so, it will send the new call to the same Asterisk box as 
the existing call.

In such a case, for each call, you need to attached to the call (as 
dialog variables) the callee, caller and the Asterisk box . When a new 
call is coming, you check if the new caller is already involved in a 
call and if so, fetch the value of the proxy in order to send to the 
same box.

For more about the technical details of the function, see
       http://www.opensips.org/html/docs/modules/devel/dialog.html#id272137

Regards,
Bogdan

-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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