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

[Bogdan-Andrei Iancu]

Hi Erik,

how does your opensips distribute the requests among the asterisk 
servers ? dispatching? or ?

Regards,
Bogdan

erik pepermans wrote:
> 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