asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	b6f4713deb	Merge security fix from 1.4 branch git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.21.2@132783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	547e232ef4	If we fail to setup the PRI request channel, don't continue, exit with an error. (closes issue #11989) Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a5f4960b3c	Fix another place where peer->callno could change at a very bad time, and also fix a place where a peer was used after the reference was released. (inspired by rev 120001) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	088ac37bfc	Save the callno when we're poking, because our peer structure could change during destruction (and thus we unlock the wrong callno, causing a cascade failure). (closes issue #12717) Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) Tested by: gewfie git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6d78e78947	Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f4cd9e3958	Revert a change made for issue #12479 . This change caused a regression such that a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's' extension anymore. (closes issue #12770) Reported by: dagmoller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	59410a3f01	Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	35ea2643d4	Even of the first PING or LAGRQ doesn't get sent because it comes up too soon, make sure to reschedule so it gets sent later. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Christian Richter	b145e2f8da	fixed compile issue when dev-mode is enabled git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Christian Richter	a3fe7673a9	Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	69213c8eb6	Change a debug message to an actual debug message git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	31d518c8e6	Merged revisions 119237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119237 \| russell \| 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) \| 7 lines - Instead of only enforcing destination call number checking on an ACK, check all full frames except for PING and LAGRQ, which may be sent by older versions too quickly to contain the destination call number. (As suggested by Tim Panton on the asterisk-dev list) - Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ from being sent before the destination call number is known. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b2f0add41c	Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e0c741c5e3	Merged revisions 119008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119008 \| russell \| 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) \| 7 lines Merge changes from team/russell/iax2-another-fix-to-the-fix As described in the following post to the asterisk-dev mailing list, only enforce destination call numbers when processing an ACK. http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html (closes issue #12631) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	65808210e9	Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	3ab48f9299	merge revision 118702 from trunk to 1.4 -- Fixes a bug in chan_iax that uses send_command to poke a peer while a channel is unlocked in some cases, and because it can cause seemingly random failures could be related to some bugs in the tracker... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	405dfcb54a	Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	a403fd8ea1	Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP. (closes issue #12501) Reported by: slimey git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	0dab692af5	Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	ac177d4459	Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	b77e30e6f5	Ensure that passed in zt_chan_conf structure is not modified in mkintf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6627976f63	Apply the autoframing setting to dialogs that do not get matched against a user or peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	e9e2df2283	Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	48f4538a8d	Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby (license 430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	915e1b570f	Avoid access of uninitialized memory. This caused a bunch of crashes for me while doing load testing of development branch where I'm working on some performance improvements. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	53061c109f	Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	4b2a679f9e	Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	59adcca238	Accept text messages even with Content-Type: text/plain;charset=Södermanländska git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	01de8fa419	Fix a deadlock involving channel autoservice and chan_local that was debugged and fixed by mmichelson and me. We observed a system that had a bunch of threads stuck in ast_autoservice_stop(). The reason these threads were waiting around is because this function waits to ensure that the channel list in the autoservice thread gets rebuilt before the stop() function returns. However, the autoservice thread was also locked, so the autoservice channel list was never getting rebuilt. The autoservice thread was stuck waiting for the channel lock on a local channel. However, the local channel was locked by a thread that was stuck in the autoservice stop function. It turned out that the issue came down to the local_queue_frame() function in chan_local. This function assumed that one of the channels passed in as an argument was locked when called. However, that was not always the case. There were multiple cases in which this channel was not locked when the function was called. We fixed up chan_local to indicate to this function whether this channel was locked or not. The previous assumption had caused local_queue_frame() to improperly return with the channel locked, where it would then never get unlocked. (closes issue #12584) (related to issue #12603) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	48dd08e321	Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	09c28afa6d	Remove debug output. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	03c5a410ad	Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 \| russell \| 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) \| 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	442079ab0b	Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f27d0168a2	Track peer references when stored in the sip_pvt struct as the peer related to a qualify ping or a subscription. This fixes some realtime related crashes. (closes issue #12588) (closes issue #12555) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e1c4c9e7b6	Merged revisions 115511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115511 \| russell \| 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) \| 3 lines Remove remnants of dlinkedlists. I didn't actually use them in the final version of my IAX2 improvements. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	fbf07008d9	Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	61bee5aa54	Add new "pri show version" command to show the libpri version for support reasons. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5f1f3ed473	Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 20000, then every media frame would cause 20000 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	26fc3d5ac6	Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	8b8a6f2486	use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	a4732cfb3c	When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	609ed327eb	Use consistent logic for checking to see if a call number has been chosen yet. Also, remove some redundant logic I recently added in a fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	709502b119	Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8eee7feb2b	Resolve a deadlock in chan_local by releasing the channel lock temporarily. (closes issue #11712) Reported by: callguy Patches: 11712.patch uploaded by putnopvut (license 60) Tested by: acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	0cd455c19b	Ensure that when we set the accountcode, it actually shows up in the CDR. (Fix for AMI Originate) (Closes issue #12007) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	57c68bcb3a	Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow up very quickly. (issue #12515) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	2acde60c29	Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	694a6b4abb	Fix find_callno_locked() to actually return the callno locked in some more cases. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	5980514bb0	Add 502 support for both directions, not only one... (see r114571) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	0c777767c9	Treat a 502 just like a 503, when it comes to processing a response code git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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4406 Commits (1.4.21.2)