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${ noResults }
23005 Commits (c3c317433f4a390c9af01d635787f78bfa34e60c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e5c20ccb76 |
Code formatting fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
7bf6a01cfa |
Fix reference leaks involving SIP Replaces transfers
The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363987 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363988 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
5746e0d2ac |
chan_sip: [general] maxforwards, not checked for a value greater than 255
The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363935 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363936 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
9d655bd0e8 |
Update Pickup application documentation. (Even better)
........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363876 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
e736a4fed3 |
* Put more information in pickup_exec() LOG_NOTICE.
* Delay duplicating a string on the stack in pickup_exec(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
0986873128 |
Update Pickup application documentation.
........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
af39a4374e |
Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call.
Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
d2ac624b87 |
Clear ISDN channel resetting state if the peer continues to use it.
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
04ddb5714f |
Add documentation
Thanks Tilghman! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
f102aecf12 |
Formatting changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
a8e755700e |
Use the DEFINED value for musicclass length.
For some reason, features.c has it's own definition. Should propably be fixed too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
7aa0c3c64b |
Make it possible to change the minimum DTMF duration in asterisk.conf
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
228ce5fd74 |
Formatting fixes
Developer guidelines are important. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
db2b162e8c |
Formatting fixes
Found a small amount of curly brackets in my hotel room here in Denmark. I hereby donate them to the Asterisk project. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
7f0dce3bd1 |
Fix recalled party B feature flags for a failed DTMF atxfer.
1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
56d10c5677 |
Hangup affected channel in error paths of bridge_call_thread().
........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363376 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
18045c9a07 |
OpenBSD doesn't have rawmemchr, use strchr
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362869 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
f663924517 |
Make app_dial and app_queue use new macro and gosub calls.
* Simplify some code in app_dial and app_queue by calling ast_app_exec_macro() and ast_app_exec_sub(). * Fix minor locking issue in app_dial for post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
f03d56a84d |
On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY.
The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363212 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363215 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
ceefcf8839 |
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) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363156 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
e8e12afc6a |
AST-2012-006: Fix crash in UPDATE handling when no channel owner exists
If Asterisk receives a SIP UPDATE request after a call has been terminated and the channel has been destroyed but before the SIP dialog has been destroyed, a condition exists where a connected line update would be attempted on a non-existing channel. This would cause Asterisk to crash. The patch resolves this by first ensuring that the SIP dialog has an owning channel before attempting a connected line update. If an UPDATE request is received and no channel is associated with the dialog, a 481 response is sent. (closes issue ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363107 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c37c7b4a2c |
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 ........ Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
eb0a8df41c |
res_corosync: Recover if corosync gets restarted.
If corosync gets restarted while Asterisk is running, automatically recover. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
41826d203c |
res_corosync: reimplement "corosync show members" command.
Reimplement the "corosync show members" CLI command using a CPG iterator instead of the cpg_membership_get API call. This will also show all CPG members, including those in groups other than 'asterisk', which may be useful at some point for debugging purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c870dad57e |
Update app_dial M and U option GOTO return value documentation.
........ Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362998 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
3a874139d4 |
Fix connected-line/redirecting interception gosubs executing more than intended.
* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so execution will stop after the routine returns there. (s@gosub_virtual_context:1) * Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and gosub application respectively with the parameter string already created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
e6d08d92e3 |
Move debug message in ast_rtp_instance_early_bridge_make_compatible().
Move debug message in ast_rtp_instance_early_bridge_make_compatible() to be output when what it states has actually happened. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
255214c5da |
Add missing payload type to events API
The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362918 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
01194c5811 |
Use ast_channel_lock_both() where it was inlined before.
The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel lock was originally obtained is overkill where ast_channel_lock_both() was inlined. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
b43f4a60dd |
* Add more information to some messages in __ast_pbx_run().
* Simplify some dialplan priority setting code in ast_explicit_goto() because of opaquification. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
34d670f786 |
Document Speech* apps hangup on failure and suggest TryExec
The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362816 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
73f48997f9 |
Add original party id and reason support.
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
92ca507d72 |
Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}.
........ Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362730 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
a011ae78dc |
Add leading and trailing backslashes
A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. ........ Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362681 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
47ccc7f5d6 |
Update membermacro and membergosub documentation in queues.conf.sample.
........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362678 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6d6bacd5cb |
Convert some strncpys to ast_copy_string
Review: https://reviewboard.asterisk.org/r/1732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
ba93541ced |
Prevent a crash in ExternalIVR when the 'S' command is sent first.
If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski ........ Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
772ad8a641 |
Handle multiple commands per connection via netconsole
Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362537 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
f78290068a |
Fix a variety of potential buffer overflows
* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
33c9161d1e |
Fix building security events test
The Security Events Framework API changed in trunk to support IPv6. This broke the building of the security events test which was based around IPv4. This patches fixes the build by changing the test to conform to the new changes. (related to issue ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c7cb03a975 |
Add ability to ignore layer 1 alarms for BRI PTMP lines.
Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
7b5eb159e9 |
Handle case where an unknown format is used to get the preferred codec size
In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
016dfa01f1 |
Fix places in resources where a negative return value could impact execution
This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362364 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f88a632d96 |
Make use of va_args more appropriate to form in various res_config modules plus utils.
A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ ........ Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362357 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3934b0478d |
Fix places in main where a negative return value could impact execution
This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362360 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2cc415417e |
Fix places where a negative return from ftello could be used as invalid input
In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362356 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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fc63e07135 |
Avoid cppcheck warnings; removing unused vars and a bit of cleanup.
Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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70c5ac6635 |
Fix error that caused seek format operations to set max file size to '1' or '0'
A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362305 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8337ecd38d |
Turn off warning message when bind address is set to any.
When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362264 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2fed9cfa8f |
Fix negative return handling in channel drivers
In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362205 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |