[asterisk-bugs] [Asterisk 0009305]: [patch] REINVITE before 200ok causes a call to be ended
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Aug 14 09:43:36 CDT 2007
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=9305
======================================================================
Reported By: atca_pres
Assigned To: oej
======================================================================
Project: Asterisk
Issue ID: 9305
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: normal
Status: confirmed
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 58957
Disclaimer on File?: Yes
Request Review:
======================================================================
Date Submitted: 03-16-2007 13:28 CDT
Last Modified: 08-14-2007 09:43 CDT
======================================================================
Summary: [patch] REINVITE before 200ok causes a call to be
ended
Description:
When Flash hooking:
Box 1 sends a INVITE with a contact 0.0.0.0 (Hold)
Asterisk sends a invite to box 2
Box 2 sends a trying
Asterisk sends a second invite to box 2 with a differetn CSeq, Branch and
Session version(SDP). This trigger a 500 msg.
The 500 is triggered because you cannot (according to RFC) send a second
invite when you have an unfinished dialog
Note : No RTP Portal
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0009142 Placing a call on hold sends two INVITE...
related to 0009209 race condition in sip hangup with reinv...
related to 0009649 BYE calls too fast when connected to a ...
======================================================================
----------------------------------------------------------------------
one47 - 08-14-07 09:43
----------------------------------------------------------------------
Yes, the state of a call is tracked in the code, but SIP is not quite that
straight-forward. An INVITE is used in several ways, and in Asterisk seems
to be tracked differently depending on how it is used.
The key mechanisms affecting your system are 1) INVITE to Set-up a call,
2) INVITE to Redirect media during a reINVITE, 3) INVITE to un-Redirect
media during a hangup.
1) was already tracked correctly.
2) is fixed by sip_reinvite4.diff
3) is still being missed.
I am uploading sip_reinvite6.diff, which if I am lucky will defer the BYE
if a media-change INVITE is outstanding, and not at other times :-O
Cheers,
Steve
Issue History
Date Modified Username Field Change
======================================================================
08-14-07 09:43 one47 Note Added: 0068829
======================================================================
More information about the asterisk-bugs
mailing list