[asterisk-users] SIP dialog matching problem? (1.4.23.1)

[Klaus Darilion]

Hi!

Actually I would consider this as a bug, thus you should report it at 
bugs.digium.com.

Are you using pedantic=yes (sip.conf)? If not, it would be interesting 
if the pedantic mode has the same problem.

regards
klaus

Santiago Gimeno schrieb:
> Hello all,
> 
> Not sure if this mail belongs to this users or dev list. Sorry about
> that.
> 
> We have the following scenario:
> 
>       PhoneA         OpenSER       Asterisk        PhoneB         PhoneC
>          |              |              |              |              |
>          |              |              |              |              |
>          |              |              |              |              |
>          |INVITE B      |              |              |              |
>          |------------->|              |              |              |
>          |              |INVITE B      |              |              |
>          |              |------------->|              |              |
>          |              |              |INVITE B      |              |
>          |              |              |------------->|              |
>          |              |              |486 Busy Here |              |
>          |              |              |<-------------|              |
>          |              |              |ACK           |              |
>          |              |              |------------->|              |
>          |              |486 Busy Here |              |              |
>          |              |<-------------|              |              |
>          |              |ACK           |              |              |
>          |              |------------->|              |              |
>          |302 MOVED (to C)             |              |              |
>          |<-------------|              |              |              |
>          |ACK           |              |              |              |
>          |------------->|              |              |              |
>          |INVITE C      |              |              |              |
>          |------------->|              |              |              |
>          |              |INVITE C      |              |              |
>          |              |------------->|              |              |
>          |              |503 Unavailable              |              |
>          |              |<-------------|              |              |
>          |              |ACK           |              |              |
>          |              |------------->|              |              |
>          |503 Unavailable              |              |              |
>          |<-------------|              |              |              |
>          |ACK           |              |              |              |
>          |------------->|              |              |              |
>          |              |              |              |              |
>          |              |              |              |              |
> 
> 
> 
> 1.- Phone A calls Phone B behind Asterisk.
> 2.- Phone B rejects call by sending a '486 Busy Here' response.
> 3.- When OpenSER receives the 486 it sends a '302 Moved Temporarily'
> to Phone A to redirect the call to Phone C.
> 4.- Phone A perfoms the redirection and sends a new INVITE to Phone C
> (that is also behind Asterisk) with same call-id BUT DIFFERENT from-tag,
> CSeq.
> 5.- Asterisk, for some reason, considers the new INVITE to belong to the
> previous call and then rejects the call with a
> '503 Unavailable'. But it cannot be considered to belong to the same
> dialog because the tags are different, although the call-id is the same.
> We have used pedantic checking. Could it be considered as a bug?
> 
> Looking at the code of chan_sip.c (version 1.4.23.1), we have observed
> that in function 'find_call' line 4667, asterisk is considering the call
> as FOUND because of this test:
> !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED).
> Commenting out this comparison, the call proceeds correctly. Sure, there
> is some reason for this checking and we would like to know which is and
> in what does it affect. How could we fix it?
> 
> The following is the asterisk console output when the call does not
> proceed:
> [Mar  2 12:15:24] DEBUG[9989]: chan_sip.c:15813 handle_request: ****
> Received INVITE (5) - Command in SIP INVITE [Mar  2 12:15:24]
> NOTICE[9989]: chan_sip.c:14724
> handle_request_invite: Unable to create/find SIP channel for this INVITE
> [Mar  2 12:15:24] DEBUG[9989]: chan_sip.c:4653 find_call: = Looking for
> Call ID: 9463D153-64F11DE-8602E9BF-A87F52E0 at 172.16.103.15 
> <mailto:9463D153-64F11DE-8602E9BF-A87F52E0 at 172.16.103.15>
> (Checking From) --From tag 182B3580-E9 --To-tag as62e21069
> 
> Any feedback would be appreciated.
> 
> Thank you in advance,
> 
> Santi
> 
> 
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