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2011-07-07Updates confbridge.conf video documentation and adds dtmf action for ↵dvossel4-16/+53
releasing video src. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326782 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Use older functions out of deference to older distrostwilson2-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326750 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Merged revisions 326689 via svnmerge from jrose1-2/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines res_odbc patch by tilghman to fix integers with null values Addresses some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as "(NULL)" rather than an actual NULL. (closes issue #1922STERISK-17791) Reported by: marcelloceschia Patches: 20110505__issue19223.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326694 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Merged revisions 326683 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326684 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Merged revisions 326681 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326682 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07live_ast: valgrind: run asterisk under valgrindtzafrir1-0/+9
Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326636 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Replace Berkeley DB with SQLite 3twilson60-760/+1108
There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326589 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Fixes newlines from being stripped from out of dialog sip MESSAGES.dvossel1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326544 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Merged revisions 326484 via svnmerge from dvossel1-28/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose discovered a performance issue with this fix that prevents his analog phones from working when using timerfd as a timing source. Until it is understood what is causing this performance problem, this patch is being reverted. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326485 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326411 via svnmerge from tilghman14-19/+19
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326412 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Prompt conversion scriptkmoore1-0/+20
Several variables in the script control which files are converted and the source and destination formats. Patch-by: Trey Blancher <support@digium.com> (closes AST-560) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326368 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326291 via svnmerge from rmudgett2-77/+217
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326321 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05New feature: AMI Action FilterAddmarkm2-19/+140
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session (closes issue ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326267 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326209 via svnmerge from mjordan1-1/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines Updated filestream destructor to block until move is complete when cache is used When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes. (closes issue ASTERISK-17724) Reported by: Adiren P. Tested by: mjordan ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326210 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-01Merged revisions 326144 via svnmerge from rmudgett1-29/+23
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326145 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-01Change CHANGES move the commits to the right placeirroot1-4/+7
r296249 r318141 Application changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326101 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-01Change CHANGES move the commits to the right place in the file missed in reviewirroot1-10/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326056 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-01updated irroots info for the authors sectionmnicholson2-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326006 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Fixes warning message caused by confbridge playback chan not being answered.dvossel1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325937 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Merged revisions 325935 via svnmerge from rmudgett2-13/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325936 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Video support for ConfBridge.dvossel11-34/+478
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Merged revisions 325877 via svnmerge from mjordan1-3/+21
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines Patched voicemail user option for emailbody / emailsubject Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325900 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Merged revisions 325821 via svnmerge from jrose1-4/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines Fixes an issue with Music on Hold classes losing files in playlist when realtime is used. The bug occurs rather intermittently and I relied on the reporters to test the patch. After a sanity check and some testing, I'm giving it an OK. (closes issue ASTERISK-17875) Reported by: David Cunningham Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325864 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38mnicholson4-81/+1131
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the dialplan. Big thanks to irroot for porting this code to use the framehooks api. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325816 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30copy all flags on asterisk frames instead of just the timing flagmnicholson1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325815 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325740 via svnmerge from kmoore2-36/+26
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325741 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325673 via svnmerge from dvossel1-0/+28
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines Fixes timerfd locking issue. (closes ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz Review: https://reviewboard.asterisk.org/r/1255/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325674 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325614 via svnmerge from rmudgett1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed error exit cleanup in app_queue.c copy_rules() and reload_queue_rules(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325616 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325610 via svnmerge from rmudgett1-1/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines Response to QueueRule manager command does not contain ActionID if it was specified. * Add ActionID support as documented for the QueueRule AMI action. * Remove documentation for ActionID with the Queues AMI action. The output does not follow normal AMI response output and there is no place to put an ActionID header. (closes issue AST-602) Reported by: Vlad Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325611 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325545 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines make framehooks prevent native bridging (for real this time) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325547 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325537 via svnmerge from mnicholson2-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines don't do native/remote bridging if a framehook is active on the channel ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325538 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Commit "distrotech" app_queue changes to Trunkirroot4-94/+290
* Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325483 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325416 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines Fix random misspelling noticed on asterisk-users. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325417 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325339 via svnmerge from dvossel1-7/+15
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325345 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325212 via svnmerge from rmudgett1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325213 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325152 via svnmerge from jrose1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines Fixes moh reload breaking custom mode moh classes when the config file is untouched (closes issue ASTERISK-17730) Reported by: sdolloff ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325153 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Fixes issue with video and text not being reinvited correctly with directmediadvossel1-6/+3
If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325151 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325091 via svnmerge from lmadsen1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line Remove line from prep_tarball that kills mkrelease. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325092 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Don't forget to build the Via when sending MESSAGEtwilson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325046 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-27Merged revisions 324955 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines Save and restore errno from within signal handlers. This is recommended by the POSIX standard, as well as by the sigaction(2) manpage for various platforms that we support (e.g. Mac OS X). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324961 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-27Merged revisions 324914 via svnmerge from rmudgett1-6/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324915 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-24Merged revisions 324849 via svnmerge from rmudgett1-35/+65
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines Syntax errors in dialplan do not display the file name. When issuing the CLI command "dialplan reload" syntax errors and warnings are displayed on the console. The offending line number is displayed on the console, but the file name is not displayed. Errors caught in main/config.c do display the file name. (closes issue ASTERISK-17985) Reported by: ulogic Patches: pbx_config.patch uploaded by ulogic (License #5685) modified format Tested by: rmudgett JIRA SWP-3554 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324850 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-24Merged revisions 324768 via svnmerge from jrose1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines DTMF wasn't being logged on connected consoles when enabled in logger.conf Previously in order for DTMF to be logged in a connected console session, the user would have to do logger set channel DTMF on. This corrects that so that it is on by default. This issue was caused by an off by one error incurred by a logger level count of 6 in logger.h where it should have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324769 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23ConfBridge: redundant code cleanupkmoore1-3/+0
There is no reason to clean up features twice. Review: https://reviewboard.asterisk.org/r/1279/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324709 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324678 via svnmerge from kmoore1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324708 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324685 via svnmerge from dvossel1-4/+12
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines Fixes sip crash when calling remove_uri_parameters with NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324689 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324652 via svnmerge from dvossel3-33/+70
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines 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: http://svn.digium.com/svn/asterisk/trunk@324664 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324557 via svnmerge from twilson1-2/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines Remove tests for parsing address with invalid port getaddrinfo on OS X returns with EAI_NONAME error when passed a port greater than 65535. Linux throws no error, so remove the tests for now. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324558 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324491 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line Use correct variable for text SRTP media. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324495 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324484 via svnmerge from twilson4-27/+251
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324487 f38db490-d61c-443f-a65b-d21fe96a405b
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