mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
19775 Commits (6f56b53c3590657de74c7bee4082fa0c2f3f044d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
6f56b53c35 |
AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization.
As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@363117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
5991853759 |
AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling
When handling a keypad button message event, the received digit is placed into a fixed length buffer that acts as a queue. When a new message event is received, the length of that buffer is not checked before placing the new digit on the end of the queue. The situation exists where sufficient keypad button message events would occur that would cause the buffer to be overrun. This patch explicitly checks that there is sufficient room in the buffer before appending a new digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@363100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
84e00510c4 |
Fix remotely exploitable stack overrun in Milliwatt
Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@359645 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
dcdc0e4c34 |
Correct two flaws in sip.conf.sample related to AST-2011-013.
* The sample file listed *two* values for the 'nat' option as being the default. Only 'yes' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@348515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
b2a75abb50 |
Update svn:externals to use menuselect from 1.6.2.20 and not later.
This change is required because when making security releases, if you pull from menuselect/trunk you'll get changes meant for later versions of Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@347659 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
d5acfbf3ff |
Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@347530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
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4da96cefd7 |
Don't set the nat default twice.
Cleaning up a small merge issue ASTERISK-18862 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@345827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
699005a0cd |
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@345800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
961a0c2314 |
fixes reference leak pointed out by rmudgett in https://reviewboard.asterisk.org/r/1337/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@330505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
356baaba35 |
Asterisk 18103 - Fix reload crash caused by destroying default parking lot
Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@330490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6d1b2a4603 |
Add .o files to svn:ignore property, since it's only ignored if locally configured to do so.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@327255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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17bfec3bca |
Merged revisions 325275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28 Jun 2011) | 2 lines Don't leak SIP username information ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@325277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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69bd6fdad7 |
Addresses AST-2011-008, memory corruption and remote crash in SIP driver.
AST-2011-008 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@324643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dc16ba95de |
Merged revisions 324627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@324634 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1f22aa216f |
ConfBridge does not handle hangup properly
When playing back a prompt to a channel, confbridge neglects to check for hangup events causing lockup condititions for hangups that occur before actually joining the conference. This change ensures that the user is removed from the conference in the event of a premature hangup. Review: https://reviewboard.asterisk.org/r/1277/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@324306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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431aff4e3e |
Merged revisions 323732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@323733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fb9e907652 |
Merged revisions 323559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@323579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fd499f1607 |
Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop.
(closes issue ASTERISK-17798) tested by mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@323039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2cd1c16b11 |
Blocked revisions 322657,322698 via svnmerge
........ r322657 | mnicholson | 2011-06-09 10:20:19 -0500 (Thu, 09 Jun 2011) | 2 lines whitespace ........ r322698 | mnicholson | 2011-06-09 10:36:05 -0500 (Thu, 09 Jun 2011) | 2 lines unlock pvt when we drop voice frames received in early media when in t.38 mode ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ffb13d7d38 |
unlock pvt when we drop voice frames received in early media when in t.38 mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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70bf4a47f4 |
fix for previous commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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52c395fb1f |
Merged revisions 322646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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91519b89c7 |
Fix *8 directed pickup locks system during pickupsound play out
move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.6.2.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@321210 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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57fb23f1aa |
Adds missing part to the ast_tcptls_server_start fails second attempt to bind patch.
(closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls_2.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d1fbf03fdf |
Fixes chanspy enforced mode lacking a channel_unlock.
(closes issue #19348) Reported by: wdoekes Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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206057bb60 |
Don't crash when the connection fails.
(closes issue #19250) Reported by: seadweller Patches: 20110514__issue19250.diff.txt uploaded by tilghman (license 14) Tested by: seadweller, sum git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8cd752f1e3 |
Blocked revisions 320393 via svnmerge
........ r320393 | pabelanger | 2011-05-21 01:09:09 -0400 (Sat, 21 May 2011) | 2 lines Solaris compatibility fixes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9764f774c8 |
Fixes issue with ast_tcptls_server_start failing on second attempt to bind.
(closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3e4e3629f3 |
Merged revisions 320235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9ed4d9d89c |
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320179 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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539b034ed6 |
Blocked revisions 320055 via svnmerge
........ r320055 | dvossel | 2011-05-20 11:38:28 -0500 (Fri, 20 May 2011) | 9 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320056 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0640d5f616 |
Merged revisions 319652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d0a4962e60 |
Merged revisions 319527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e59b9085d7 |
Unlink a peer from peers_by_ip when expiring a registration
Review: https://reviewboard.asterisk.org/r/1218/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d06dd1262f |
Fixes issue with peer ref-counting during handle_request_subscribe.
(closes issue #19293) Reported by: irroot git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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229e1541be |
Make sure tcptls_session exists before dereferencing it.
(closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9172658d42 |
Merged revisions 318734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d12154f767 |
Merged revision 222981 from
https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar deadlock possible when running the Pickup application internally. ------------------------------------------------------------------------ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009) | 13 lines Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fe41f00a47 |
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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12e2f9cdeb |
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d693f81ab6 |
Fixes cases where sip_set_rtp_peer can return too early during media path reset.
(closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7eca34890a |
pbx_lua autoservice fixes
Don't start an autoservice in pbx_lua if pbx_lua already started one and don't stop one if we didn't start one. Also start and stop the autoservice when transferring control from and to the pbx. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5081b488fe |
Merged revisions 317719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011) | 11 lines Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) Uninitialized variable. (issue #18640) (closes issue #18785) Reported by: pnlarsson Patches: issue18785_enegaard.patch uploaded by enegaard (license 1197) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c4ed0bac68 |
Add a datastore fixup to fix a pbx_lua crash.
(closes issue #19055) Reported by: jamhed Patches: lua_datastore_fixup1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, jamhed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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30e8e7dedb |
Merged revisions 317574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5fa33b9e28 |
Merged revisions 317211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b9bf3d604e |
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5731697675 |
Merged revisions 317102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) | 8 lines Disable console colourization inside safe_asterisk checks. (closes issue #19213) Reported by: lefoyer Patches: issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717) Tested by: wdoekes, lefoyer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ad0b87f72b |
Merged revisions 316707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1ef5452cd8 |
Fixes one-way-audio when chanspy activated with the 'o' option
(closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |