13 Commits (914c8e28c1bb99b8217f93b51ce8f59870350fb6)

Author	SHA1	Message	Date
Richard Mudgett	69810b306d	chan_sip.c: Fix autokillid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f	9 years ago
Richard Mudgett	cb97198ca6	chan_sip.c: Simplify sip_pvt destructor call levels. ... Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a	9 years ago
Corey Farrell	d486659502	chan_sip: Simplify dialog/peer references, improve REF_DEBUG output. ... * Replace functions for ref/undef of dialogs and peers with macro's to call ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller information to sip_alloc and find_call. ASTERISK-24882 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4189/ ........ Merged revisions 433115 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. ... Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Jonathan Rose	cd37bec058	logger: Adds additional support for call id logging and chan_sip specific stuff ... This patch improves the handling of call id logging significantly with regard to transfers and adding APIs to better handle specific aspects of logging. Also, changes have been made to chan_sip in order to better handle the creation of callids and to enable the monitor thread to bind itself to a particular call id when a dialog is determined to be related to a callid. It then unbinds itself before returning to normal monitoring. review: https://reviewboard.asterisk.org/r/1886/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	0ccc1f5274	Merged revisions 353321 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	2b2d34b3c9	Constify tag argument in REF_DEBUG related code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b63c1cc545	Fix some potential deadlocks pointed out by helgrind. ... * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b42c6cab17	Revert early destruction of RTP sessions ... Some code improperly assumes that the sessions are still there, so revert the change until I can find all of them and fix them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	cb160a12b0	Destroy RTP fds when we schedule final dialog destruction ... Since we are only keeping the dialog around for retransmissions at this point and there is no possibility that we are still handling RTP, go ahead and destroy the RTP sessions. Keeping them alive for 32 past when they are used is unnecessary and can lead to problems with having too many open file descriptors, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. ... This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a1fe641a38	retransmit response to BYE requests until timer J expires ... According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T1*64 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e7a5fb5459	Make all of the various rtpqos parameters in this branch available from the CHANNEL function. ... Also includes a test for retrieving rtpqos parameters, including a NULL RTP driver. Additionally, some further separation of the SIP internal API into headers was necessary. (closes issue #16652) Reported by: kkm Patches: 20100204__issue16652.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/501/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago