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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@212958 ↵v1.4.11kpfleming8-20/+19
f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21use autotagged externalsrussell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80193 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21importing files for 1.4.11 releaserussell3-0/+10851
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80192 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Creating tag for the release of asterisk-1.4.11russell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80191 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Creating tag for the release of asterisk-1.4.11russell3-10851/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80189 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21use autotagged externalsrussell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80187 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21importing files for 1.4.11 releaserussell3-0/+10851
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80186 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Creating tag for the release of asterisk-1.4.11russell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.11@80185 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Don't record SIP dialog history if it's not turned on. Also, put an upperrussell1-1/+20
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) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21ugh. removing the diffs from ulaw.h and alaw.h for now; accidentally added ↵murf2-99/+1
them in 80166 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21This patch solves problem 1 in 8126; it should not slow down the alaw codec, ↵murf3-2/+100
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Don't try to dereference the owner channel when it may not existrussell1-10/+7
(issue #10507, maxper) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80132 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21(issue #10510)qwell1-17/+15
Reported by: casper Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few errors in sample cdr config file. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80130 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20Fix the build of app_queuerussell1-49/+49
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80088 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20After a discussion on #asterisk-dev, it was decided that this should be in ↵mmichelson1-0/+2
1.4 as well. (issue #10424, reported and patched by irroot) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80086 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20Found a pointless ternary if. member->dynamic was set to 1 and has no ↵mmichelson1-1/+1
opportunity to change between then and this line, so "dynamic" will ALWAYS be output. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80049 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20(issue #10499)qwell1-8/+8
Reported by: casper Patches: extensions.conf.sample.diff uploaded by casper (license 55) Update CLI examples in extensions.conf.sample to reflect command changes. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80047 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20Ukrainian language voicemail support.mmichelson1-0/+95
(closes issue #10458, reported and patched by Oleh) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20Missing curly braces. Oops. (Reported by snuffy via IRC)tilghman1-3/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79998 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-18Don't allocate vmu for messagecount when we could just use the stack instead ↵tilghman1-15/+2
(closes issue #10490) Also, remove a useless (and leaky) SQLAllocHandle (closes issue #10480) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79947 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17Avoid a crash in the handling of DTMF based Caller ID. It is valid forrussell1-0/+2
ast_read to return NULL in the case that the channel has been hung up. (crash reported by anonymouz666 on IRC in #asterisk-dev) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79912 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17Patch allows for more seamless transition from file storage voicemail to ↵mmichelson1-1/+11
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79906 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17Don't send a semicolon over the wire in sip notify messages.qwell3-1/+23
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). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79904 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17Re-add the setting of callerid name and number.qwell1-0/+6
Issue 10485, reported by and fix explained by paradise. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79902 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17Fix some crashes in chan_sip. This patch changes various places that add itemsrussell1-2/+13
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) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79857 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-17sometimes we don't need to signal dtmf tones to asterisk, we just want them ↵crichter1-4/+5
to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79833 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Fix a little race condition that could cause a crash if two channels had MOHrussell1-3/+1
stopped at the same time that were using a class that had been marked for deletion when its use count hits zero. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79792 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16This patch fixes a bug where reloading the module with "module reload" did notrussell1-30/+48
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79778 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Fix more deadlocks in chan_iax2 that were introduced by making frame handlingrussell1-12/+164
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 function holding its lock due to calling a function that has to unlock/lock it to avoid deadlocks. I went through and fixed all of these places to account for this possibility. (issue #10362, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79756 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Fixes a problem where agents would get stuck busy due to their wrapuptime ↵mmichelson1-0/+1
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! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79748 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16base_encode is not trying to open a log file, so we should not call it a log ↵mmichelson1-1/+1
file in the warning. (related to issue #10452, reported by bcnit) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79690 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16A fix for two critical problems detected while working with Danielphsultan1-0/+27
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 trying to access the found->cap pointer if the resource list for the given buddy is not empty. This situation is perfectly valid and must be handled. The Gizmoproject's jabber client put that problem in evidence. Also added a few comments in the code as well as a handle for the capabilities from Gmail's jabber client, which are stored in a caps:c tag rather than the usual c tag. Closes issue #10184. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79665 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-160x80 + protocol is wrong for USERUSER when we want to send IA5 Chars.crichter1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79642 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-15(closes issue #10440)file1-1/+1
Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79553 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-15Fixed an error in the Russian language voicemail intro.mmichelson1-1/+1
(issue #10458, reported and patched by Oleh) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79527 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-15(closes issue #10456)file1-1/+1
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79523 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-14Fix another spot where an iax2_peer would be leaked if realtime was in use.russell1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79470 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-14Fix some memory leaks throughout chan_iax2 related to the use of realtime.russell1-3/+15
I found these while working on iax2_peer object reference tracking. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79436 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-14(closes issue #10415)file1-7/+11
Reported by: atis Revert fix for #10327 as it causes more issues then it solves. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79397 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13memset really, really needs to be used here.murf1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79363 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Instead of accepting a single DTMF character accept a full string.file3-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79334 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Don't call find_peer in registry_authrequest with the pvt lock held to avoid arussell1-5/+17
deadlock. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79301 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Release the pvt lock before calling find_peer in register_verify to avoid arussell1-6/+13
deadlock. Also, remove some unnecessary locking in auth_fail that was only done recursively. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79276 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Don't call find_peer within update_registry with a pvt lock held. This canrussell1-6/+22
cause a deadlock as the code will eventually call find_callno. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79274 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13I am fighting deadlocks in chan_iax2. I have tracked them down to a singlerussell1-2/+28
core issue. You can not call find_callno() while holding a pvt lock as this function has to lock another (every) other pvt lock. Doing so can lead to a classic deadlock. So, I am tracking down all of the code paths where this can happen and fixing them. The fix I committed earlier today was along the same theme. This patch fixes some code down the path of authenticate_reply. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79272 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13This patch fixes bug 10411. I added a new regression test, some regression ↵murf10-129/+204
test cleanups git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79255 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Fix a potential deadlock in socket_process. check_provisioning can eventuallyrussell1-2/+16
call find_callno. You can't hold a pvt lock while calling find_callno because it goes through and locks every single one looking for a match. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79214 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Add an API call to allow the engine to know that DTMF was received.file3-0/+20
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13(closes issue #10437)file11-29/+1
Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79174 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-11Ensure the connection gets marked as used at allocation time (closes issue ↵tilghman1-2/+10
#10429, report and fix by mnicholson) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79142 f38db490-d61c-443f-a65b-d21fe96a405b
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