Santiago Gimeno
2009-Mar-02 12:40 UTC
[asterisk-users] SIP dialog matching problem? (1.4.23.1)
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
(Checking From) --From tag 182B3580-E9 --To-tag as62e21069
Any feedback would be appreciated.
Thank you in advance,
Santi
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Klaus Darilion
2009-Mar-04 08:35 UTC
[asterisk-users] SIP dialog matching problem? (1.4.23.1)
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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users