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

Tue May 11 13:15:13 CEST 2010

Hi Bogdan,

I presume the first BYE to OpenSIPS should be forwarded to 11.01.01.001 ?

What do you suggest as a solution ?

U 2010/05/06 22:50:01.372924 22.02.02.002:5060 -> 192.168.5.1:5071
BYE sip:0479984159 at 11.01.01.001 SIP/2.0.
Via: SIP/2.0/UDP 22.02.02.002:5060;branch=z9hG4bK09456547;rport.
Route: <sip:192.168.5.1:5071;lr=on;did=244.7d4366b6>.
From: <sip:0479984159 at 192.168.5.1:5071>;tag=as467b1d3e.
To: "0479984159" <sip:0479984159 at foo.domain.com>;tag=as1a9a691f.
Call-ID: 09820d5a02b9bfe520c2c1017dbcf0b8 at foo.domain.com.
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.

#
U 2010/05/06 22:50:01.390636 192.168.5.1:5071 -> 22.02.02.002:5060
BYE sip:0479984159 at 22.02.02.002:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.5.1:5071;branch=z9hG4bK1133.551ea136.0.
Via: SIP/2.0/UDP
22.02.02.002:5060;received=22.02.02.002;branch=z9hG4bK09456547;rport=5060.
From: <sip:0479984159 at 192.168.5.1:5071>;tag=as467b1d3e.
To: "0479984159" <sip:0479984159 at foo.domain.com>;tag=as1a9a691f.
Call-ID: 09820d5a02b9bfe520c2c1017dbcf0b8 at foo.domain.com.
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.

Brgds
Erik

-----Oorspronkelijk bericht-----
Van: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Namens erik pepermans
Verzonden: maandag 10 mei 2010 23:11
Aan: 'OpenSIPS users mailling list'
Onderwerp: Re: [OpenSIPS-Users] [NEW] exchanging info between dialogs

Hi Bogdan,

I included an ngrep sip trace file which should explain :

2 asterisk servers on public addresses 11.01.01.001 and 22.02.02.002 are
connected to OpenSIPS which listens on port 5071 at private address
192.168.5.1 (public address 99.09.09.009)

- 11.01.01.001 sends an invite to 192.168.5.1 with the phone number of A
- 192.168.5.1 prefixes the phone number of A via drouting module
- 192.168.5.1 sends an invite to 22.02.02.002 with the prefixed phone number
of A
- 22.02.02.002 dials A
- A picks up the phone and dials the phone number of B
- 22.02.02.002 sends an invite to 192.168.5.1 with the phone number of B
- 192.168.5.1 prefixes the phone number of B via drouting module
- 192.168.5.1 sends an invite to 22.02.02.002 with the prefixed phone number
of B
- 22.02.02.002 dials B
- B picks up the phone and talks to A

So far so good; both legs have their own dialog did; then

- B hangs up
- 22.02.02.002 sends BYE to 192.168.5.1
- 192.168.5.1 sends BYE to 22.02.02.002 where I expect it should go to
11.01.01.001

11.01.01.001 never receives a BYE and keeps the session live forever -

Thanks for your effort -

Brgds
Erik

-----Oorspronkelijk bericht-----
Van: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Namens Bogdan-Andrei Iancu
Verzonden: maandag 10 mei 2010 14:47
Aan: OpenSIPS users mailling list
Onderwerp: Re: [OpenSIPS-Users] [NEW] exchanging info between dialogs

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

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