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2008-07-22Fixes for AST-2008-010 and AST-2008-011tilghman1-1/+28
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@132711 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-03Copy the From header into a variable so that pedantic SIP handling does not ↵file1-5/+6
try to mess with a NULL pointer. (AST-2008-008) (closes issue #12607) Reported by: hooi git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@120109 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-30- Instead of only enforcing destination call number checking on an ACK, checkrussell1-11/+53
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: http://svn.digium.com/svn/asterisk/branches/1.2@119237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-29Merge changes from team/russell/iax2-another-fix-to-the-fixrussell1-7/+8
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@119008 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-08Fix a race condition that bbryant just found while doing some IAX2 testing.russell1-1/+33
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: http://svn.digium.com/svn/asterisk/branches/1.2@115564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-07Remove remnants of dlinkedlists. I didn't actually use them in the final ↵russell1-3/+0
version of my IAX2 improvements. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@115511 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05Merge changes from team/russell/iax2_find_callno_1.2russell1-134/+255
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: http://svn.digium.com/svn/asterisk/branches/1.2@115296 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-22When we receive a full frame that is supposed to contain our call number,russell1-17/+34
ensure that it has the correct one. (closes issue #10078) (AST-2008-006) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@114561 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20Fix some very broken code that was introduced in 1.2.26 as a part of the ↵russell2-14/+9
security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@110335 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Do not return with a successful authentication if the From header ends up empty.qwell1-2/+0
(AST-2008-003) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@109391 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-23Fix for fix for security fix (third time's the charm?)tilghman1-25/+24
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@94661 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-20Fix another potential seg fault ...russell1-2/+2
(closes issue #11606) Reported by: dimas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@94255 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-20Fix a couple of places where it's possible to dereference a NULL pointer.russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@94214 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Oops, missed this one casetilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@93675 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Fixing AST-2007-027 (Closes issue #11119)tilghman2-10/+95
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@93667 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-07Revert patch committed for issue #9660. It broke E&M trunks.russell1-9/+3
(closes issue #10360) (closes issue #10364) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@78370 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-25this fixes bug 10293, where the error message because defaultzone or ↵murf1-1/+1
loadzone was not defined was confusing git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76978 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-24Don't create the Asterisk channel until we are starting the PBX on it.qwell1-16/+14
(ASA-2007-018) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76802 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-23(closes issue #5866)file1-2/+2
Reported by: tyler Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76653 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-23(closes issue #10236)file1-0/+5
Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76560 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21Backport a fix for a memory leak that was fixed in trunk in reivision 76221russell1-0/+1
by rizzo. The memory used for the localaddr list was not freed during a configuration reload. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76226 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-20(closes issue #10247)file1-0/+3
Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76080 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19When processing full frames, take sequence number wraparound into account whenrussell1-1/+3
deciding whether or not we need to request retransmissions by sending a VNAK. This code could cause VNAKs to be sent erroneously in some cases, and to not be sent in other cases when it should have been. (closes issue #10237, reported and patched by mihai) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75927 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-18When traversing the queue of frames for possible retransmission afterrussell1-1/+1
receiving a VNAK, handle sequence number wraparound so that all frames that should be retransmitted actually do get retransmitted. (issue #10227, reported and patched by mihai) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75757 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17Properly check for the length in the skinny packet to prevent an invalid memcpy.russell1-1/+1
(ASA-2007-016) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75449 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17Ensure that when encoding the contents of an ast_frame into an iax_frame, thatrussell3-4/+17
the size of the destination buffer is known in the iax_frame so that code won't write past the end of the allocated buffer when sending outgoing frames. (ASA-2007-014) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75444 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17After parsing information elements in IAX frames, set the data length to zero,russell1-0/+1
so that code later on does not think it has data to copy. (ASA-2007-015) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75440 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-13(closes issue #9660)russell1-3/+12
Reported by: mmacvicar Patches submitted by: bbryant, russell Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous When using a TDM400P (and probably other analog cards) there was a chance that you could hang up and pick the phone back up where it has been long enough to be not considered a flash hook, but too soon such that the device reports that it is busy and the person on the phone will only hear silence. This patch makes chan_zap more tolerant of this and gives the device a couple of seconds to succeed so the person on the phone happily gets their dialtone. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@75052 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11The function make_trunk() can fail and return -1 instead of a valid new callrussell1-3/+8
number. Fix the uses of this function to handle this instead of treating it as the new call number. This would cause a deadlock and memory corruption. (possible cause of issue #9614 and others, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@74766 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11The cli command "agent logoff Agent/x soft" did not work...at all. Now it does.mmichelson1-0/+2
(closes issue #10178, reported and patched by makoto, with slight modification for 1.4 and trunk by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@74719 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11Use some Makefile magic to determine if linux/compiler.h is present. (issue ↵file2-0/+7
#10174 reported by francesco_r) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@74587 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-10Fix an issue with wrapuptime not working when using AgentLogin.qwell1-4/+6
Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@74376 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-09Several chan_zap options were not working on reload because they were ↵qwell1-8/+8
arbitrarily disallowed when reloading some/most PRI options (such as signalling) was disallowed. Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload. This corrects that behavior. Issue 9186, patch by tzafrir. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@74158 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06If a sip_pvt struct has already registered an extension state callback,russell1-1/+4
remove the old one before adding a new one. If this isn't done, Asterisk will crash. (issue #10120) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06(closes issue #10125)russell1-2/+6
Reported by: makoto Patches submitted by: makoto This fixes a crash in chan_sip that happens when the bindaddr setting is not valid on Asterisk startup, gets fixed, and then a reload gets issued. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73678 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06Fixed a bug wherein agents get stuck busy. (issue 9618, reported by ↵mmichelson1-1/+1
jiddings, patched by moi) closes issue #9618 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73674 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support ↵kpfleming1-0/+3
it for G.729 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73547 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05Copy language information to the dialog structure when calling a peer for ↵file1-0/+2
situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05Actually check to make sure a PBX was started on one of the Local channels ↵file1-3/+3
instead of blindly assuming it was. (issue #10112 reported by makoto) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73318 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-04bchannel configurations like echocancel and volume control, need to be ↵crichter1-0/+20
setuped on inbound calls too. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73252 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-04bad bug in overlapdial case, we called start_pbx multiple times, because the ↵crichter1-1/+6
state wasn't changed.. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-03fixed issue, that misdn_l2l1_check could only be called from mISDN Source ↵crichter1-5/+0
channels.. #9449 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73004 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-29Backport changes that make chan_iax2 not start the PBX on an incoming channelrussell1-8/+34
until the three-way call setup is completed. These changes are already in 1.4 and trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72629 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-29check if the bchannel stack id is already used, if so don't use it a second ↵crichter2-1/+17
time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72585 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27simplified generation for dummy bchannels, also we mark them as dummies, so ↵crichter4-33/+45
they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72099 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27simplified channel finding and locking a lot. removed unnecessary #ifdefed ↵crichter2-91/+67
areas. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72087 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27isdn_lib.c didn't compilecrichter1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72041 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27for inbound TE calls, we setup the bchannel when we get the ↵crichter1-45/+4
CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. removed some #if 0 areas which weren't used anymore. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@72040 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25Ignore other URIs after the first in a 300 Multiple Choice response. (issue ↵file1-1/+3
#10041 reported by homesick) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@71414 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21we activate the bchannels in TE mode on incoming calls only when we want to ↵crichter2-1/+2
connect the call. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@70672 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-28 03:22:45 +0000