Commits

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Author Commit Message Labels Comments Date
russell
Don't record SIP dialog history if it's not turned on. Also, put an upper limit on how many history entires will be stored for each SIP dialog. It is currently set to 50, but can be increased if deemed necessary. (closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer, patches updated by me) (Security implications documented in AST-2007-020)
Tags
1.4.11
Branches
1.4
murf
ugh. removing the diffs from ulaw.h and alaw.h for now; accidentally added them in 80166
Branches
1.4
murf
This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation.
Branches
1.4
russell
Don't try to dereference the owner channel when it may not exist (issue #10507, maxper)
Branches
1.4
qwell
(issue #10510) Reported by: casper Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few errors in sample cdr config file.
Branches
1.4
russell
Fix the build of app_queue
Branches
1.4
mmichelson
After a discussion on #asterisk-dev, it was decided that this should be in 1.4 as well. (issue #10424, reported and patched by irroot)
Branches
1.4
mmichelson
Found a pointless ternary if. member->dynamic was set to 1 and has no opportunity to change between then and this line, so "dynamic" will ALWAYS be output.
Branches
1.4
qwell
(issue #10499) Reported by: casper Patches: extensions.conf.sample.diff uploaded by casper (license 55) Update CLI examples in extensions.conf.sample to reflect command changes.
Branches
1.4
mmichelson
Ukrainian language voicemail support. (closes issue #10458, reported and patched by Oleh)
Branches
1.4
tilghman
Missing curly braces. Oops. (Reported by snuffy via IRC)
Branches
1.4
tilghman
Don't allocate vmu for messagecount when we could just use the stack instead (closes issue #10490) Also, remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
Branches
1.4
russell
Avoid a crash in the handling of DTMF based Caller ID. It is valid for ast_read to return NULL in the case that the channel has been hung up. (crash reported by anonymouz666 on IRC in #asterisk-dev)
Branches
1.4
mmichelson
Patch allows for more seamless transition from file storage voicemail to ODBC storage voicemail. If a retrieval of a greeting from the database fails, but the file is found on the file system, then we go ahead an insert the greeting into the database. The result of this is that people who switch from file storage to ODBC storage do not need to rerecord their voicemail greetings.
Branches
1.4
qwell
Don't send a semicolon over the wire in sip notify messages. Caused by fix for issue 9938. I basically took the code that existed before 9938 was fixed, and copied it into a new function - ast_unescape_semicolon There should be very few places this will be needed (pbx_config does NOT need this (see issue 9938 for details)) Issue 10430, patch by me, with help/ideas from murf (thanks murf).
Branches
1.4
qwell
Re-add the setting of callerid name and number. Issue 10485, reported by and fix explained by paradise.
Branches
1.4
russell
Fix some crashes in chan_sip. This patch changes various places that add items to the scheduler to ensure that they don't overwrite the ID of a previously scheduled item. If there is one, it should be removed. (closes issue #10391, closes issue #10256, probably others, patch by me)
Branches
1.4
crichter
sometimes we don't need to signal dtmf tones to asterisk, we just want them to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone.
Branches
1.4
russell
Fix a little race condition that could cause a crash if two channels had MOH stopped at the same time that were using a class that had been marked for deletion when its use count hits zero.
Branches
1.4
russell
This patch fixes a bug where reloading the module with "module reload" did not delete classes from memory that were no longer in the config. This patch fixes that problem as well as another one. Previously, if you reloaded MOH using the "moh reload" CLI command, which behaved differently than "module reload ...", MOH had to be stopped on every channel and started again immediately. However, there was no way to tell what class was being used, so they would all fall back to the default class. (closes issue #10139) Reported by: blitzrage Patches: asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich
Branches
1.4
russell
Fix more deadlocks in chan_iax2 that were introduced by making frame handling and scheduling multi-threaded. Unfortunately, we have to do some expensive deadlock avoidance when queueing frames on to the ast_channel owner of the IAX2 pvt struct. This was already handled for regular frames, but ast_queue_hangup and ast_queue_control were still used directly. Making these changes introduced even more places where the IAX2 pvt struct can disappear in the context of a functio…
Branches
1.4
mmichelson
Fixes a problem where agents would get stuck busy due to their wrapuptime being longer than the queue's wrapuptime and ringinuse=no for the queue. (closes issue #10215, reported by Doug, repaired by me) Special thanks to fkasumovic for pointing out the source of the problem and to bweschke for helping to come up with a solution!
Branches
1.4
mmichelson
base_encode is not trying to open a log file, so we should not call it a log file in the warning. (related to issue #10452, reported by bcnit)
Branches
1.4
phsultan
A fix for two critical problems detected while working with Daniel McKeehan in issue #10184. Upon priority change, the resource list is not NULL terminated when moving an item to the end of the list. This makes Asterisk endlessy loop whenever it needs to read the list. Jids with different resource and priority values, like in Gmail's and GoogleTalk's jabber clients put that problem in evidence. Upon reception of a 'from' attribute with an empty resource string, Asterisk crashes when tryi…
Branches
1.4
crichter
0x80 + protocol is wrong for USERUSER when we want to send IA5 Chars.
Branches
1.4
file
(closes issue #10440) Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low.
Branches
1.4
mmichelson
Fixed an error in the Russian language voicemail intro. (issue #10458, reported and patched by Oleh)
Branches
1.4
file
(closes issue #10456) Reported by: irroot Patches: sip_timeout.patch uploaded by irroot (license 52) Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.
Branches
1.4
russell
Fix another spot where an iax2_peer would be leaked if realtime was in use.
Branches
1.4
russell
Fix some memory leaks throughout chan_iax2 related to the use of realtime. I found these while working on iax2_peer object reference tracking.
Branches
1.4
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