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2012-01-24Set core sounds version to 1.4.22.Jonathan Rose
Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey ........ Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352373 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Opaquify channel stringfieldsTerry Wilson
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Fix locking issues with channel datastores in func_odbc.c.Richard Mudgett
* Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ ........ Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352292 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Blocked revisions 352288Joshua Colp
........ Blocked revisions 352287 ........ Move RTP timeout check to before bridged channel check so it is actually executed. (issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534) Reported by: kriborgen Patches: chan_sip.patch uploaded by kriborgen (license 6138) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Fix grammar of comment.Mark Michelson
........ Merged revisions 352230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352231 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Fix blind transfers from failing if an 'h' extension is present.Mark Michelson
This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 ........ Merged revisions 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352228 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layerMatthew Jordan
While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) ........ Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352149 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Add an announcement option to music-on-hold - plays sound when put on ↵Jonathan Rose
hold/between songs This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Adds the ability to stop specific mixmonitors by using unique IDs set at ↵Jonathan Rose
monitor launch. MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list <channel>" which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-23Fix sip_cfg.notifycid to be set with the defined enum values.Richard Mudgett
The invalid value used when notifycid was enabled was benign. As far as the code was concerned -1 and 1 are equivalent. (closes issue ASTERISK-19232) Reported by: Eike Kuiper ........ Merged revisions 352090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352091 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-21Fix ast_app_dtget() time unit inconsistency.Richard Mudgett
Note: Noone calls ast_app_dtget() with the timeout parameter of zero so the bad code normally will never get executed. * Fix unnecessary floating point division in func_timeout.c timeout_write() when all other values are integers. (closes issue ASTERISK-16817) Reported by: Dmitry Andrianov ........ Merged revisions 352029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352035 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-21Remove XXX comment that is not necessary.Mark Michelson
........ Merged revisions 352016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352017 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-21Fix RTP reference leak.Mark Michelson
If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference for the RTP instance of the transferer. This fixes the issue by merging two similar but slightly conflicting sections of code into a single area. It also adds a stop_media_flows() call in the case that the transferer's UA never sends a BYE to us like it is supposed to. (issue ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ ........ Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352015 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Make CLI sip show channel list the complete route set.Richard Mudgett
(closes issue ASTERISK-16877) Reported by: klaus3000 Patches: show-complete-routeset-patch.txt (license #5054) patch uploaded by klaus3000 (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20SIP session timeout AMI eventKinsey Moore
Add an AMI event in the Call category that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration. Event description: Event: SessionTimeout Source: source Channel: channel-name Uniqueid: channel-unique-id `source` can be either RTPTimeout or SIPSessionTimer (closes issue ASTERISK-16467) Patch-by: Kirill Katsnelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Various parking improvements.Mark Michelson
* Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Prevent potential buffer overflow on AMI MixMonitor command.Mark Michelson
Don't be alarmed. This only affected trunk, and it would have required manager access to your system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20More corrections for the ilbc codeKinsey Moore
These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351861 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Restore LSF_check function calls from set/unused variable removalKinsey Moore
These functions are not noops and modify the array that is passed in. Thanks for the catch Richard. ........ Merged revisions 351818 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Remove more set, but unused variables in the ilbc codecKinsey Moore
GCC 4.6.3 caught these in dev mode as well. ........ Merged revisions 351816 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Adds setting of mwi_from field to check_auth_result check_peer_okJonathan Rose
(closes ASTERISK-19057) Reported By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license 5242) ........ Merged revisions 351759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351762 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Remove unused variable 'tmp' from helpfun in ilbc codecMatthew Jordan
gcc version 4.6.2 caught an unused variable in the ilbc codec library. This would prevent compilation with --enable-dev-mode; variable removed. ........ Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351761 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20enable doxygen build for files in the channels/sip folder like reqresp_parser.cStefan Schmidt
........ Merged revisions 351707 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351708 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Misc minor fixes in reqresp_parser.c and chan_sip.c.Richard Mudgett
* Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351646 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Correct output of RTCP jitter statistics in SR and RR reportsKinsey Moore
Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) ........ Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351612 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Eliminates doubling the :port part of SIP Notify Message-Account headers.Jonathan Rose
This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351560 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Prevent crash when an SDP offer is received with an encrypted video stream ↵Joshua Colp
when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda ........ Merged revisions 351504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351505 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-18Include iLBC source code for distribution with AsteriskMatthew Jordan
This patch includes the iLBC source code for distribution with Asterisk. Clarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder. Review: https://reviewboard.asterisk.org/r/1675 Review: https://reviewboard.asterisk.org/r/1649 (closes issue: ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan ........ Merged revisions 351450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351451 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-18The get_pai function in chan_sip.c didn't recognized a proper callerid name andStefan Schmidt
number from a P-Asserted-Identity cause the header parsing logic was wrong. Changing the parsing functions to the sip header parsing APIs in reqresp_parser.h solves this problem. Review: https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and Mark Michelson ........ Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351408 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Fix support for parallel building with make (-j).Walter Doekes
Previously make -j <N> would cause a race between doing cleanup of certain files (defaults.h, menuselect, ...) and creating them anew. Add a new target that depends on cleanup only and has a submake doing the rest as command string. This way the cleanup goes first. (closes issue ASTERISK-18751) Tested by: Jeremy Kister Reviewed by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1660 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Eliminate odd initialization of probation variable.Mark Michelson
........ Merged revisions 351306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351308 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Adds pjmedia probation concepts to res_rtp_asterisk's learning mode.Jonathan Rose
In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Use built-in parsing functions for Contact and Record-Route headers.Mark Michelson
If a Contact or a Record-Route header had a quoted string with an item in angle brackets, then we would mis-parse it. For instance, "Bob <1234>" <1234@example.org> would be misparsed as having the URI "1234" The fix for this is to use parsing functions from reqresp_parser.h since they are heavily tested and are awesome. (issue ASTERISK-18990) ........ Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351286 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Fix udptl issue with initial INVITE introduced by r351027Matthew Jordan
When an inital INVITE occurs that contains image media, a channel is not yet associated with the SIP dialog. The file descriptor associated with the udptl session needs to be set in initialize_udptl or in sip_new to account for this scenario. ........ Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351234 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Merged revisions 351183 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines Add some missing locking in chan_sip. This patch adds some missing locking to the function send_provisional_keepalive_full(). This function is called from the scheduler, which is processed in the SIP monitor thread. The associated channel (or pbx) thread will also be using the same sip_pvt and ast_channel so locking must be used. The sip_pvt_lock_full() function is used to ensure proper locking order in a safe manner. In passing, document a suspected reference counting error in this function. The "fix" is left commented out because when the "fix" is present, crashes occur. My theory is that fixing it is exposing a reference counting error elsewhere, but I don't know where. (Or my analysis of this being a problem could have been completely wrong in the first place). Leave the comment in the code for so that someone may investigate it again in the future. Also add a bit of doxygen to transmit_provisional_response(). (closes issue ASTERISK-18979) Review: https://reviewboard.asterisk.org/r/1648 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Ensure ACK retransmit & hangup on non-200 response to INVITETerry Wilson
When handling a non-2xx final response on an INVITE transaction, we have to keep the transaction around after we send an ACK in case we receive a retransmission of the response so we can re-transmit the ACK, but also tear down the ast_channel as soon as we transmit the ACK. Before this patch, we could fail at both of these things. Calling sip_alreadygone/needdestroy prevented us from keeping the transaction up and retransmitting the ACK, and queueing CONGESTION was not sufficient to cause the channel to be torn down when originating calls via the CLI, for example. This patch queues a hangup with CONGESTION instead of just queueing CONGESTION for these responses and removes the sip_alreadygone and sip_needdestroy calls from handle_response_invite on non-2xx responses. It relies on the hangup calling sip_scheddestroy. For more information, see section 17.1.1.1 of RFC 3261. (closes issue ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ ........ Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351131 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Don't prematurely stop SIP session timerTerry Wilson
When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry. (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: session_timer_fix.diff by Terry Wilson (License #5357) based on session_timer.patch by Thomas Arimont (License #5525) ........ Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351081 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Add ABS() absolute value function to the expression parser.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Create and initialize udptl only when dialog negotiates for image mediaMatthew Jordan
Prior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or what an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This occurred even in non-INVITE dialogs that would never send image media. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates through the appropriate control frame that a dialog is to support T.38. (closes issue ASTERISK-16698) Reported by: under Tested by: Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar Broad Tested by: Stefan Schmidt review: https://reviewboard.asterisk.org/r/1668/ ........ Merged revisions 351027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351028 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Sort the output of 'database showkey' as well.Sean Bright
You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. ........ Merged revisions 350978 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Add missing code to set direct RTP setup information during dialing.Joshua Colp
........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Sort the output of 'database show' by key.Sean Bright
This more closely mimics the behavior of 'database show' before the conversion to sqlite3. ........ Merged revisions 350938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-15Allow only one thread at a time to do asterisk cleanup/shutdown.Walter Doekes
Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350889 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-15Fix -Werror=unused-but-set-variable compile error in utils/extconf.c.Walter Doekes
Note that I'm not confirming legitimacy of having that file in tree at all. Is anyone using aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged revisions 350885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350886 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-14Ensure that all AC_LANG_PROGRAM calls in the configure script are properly ↵Kevin P. Fleming
quoted. Recent versions of autoconf (2.68 on my system) won't properly process the configure script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script were, but many were not. This patch corrects the unquoted calls. ........ Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350838 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-14Multiple revisions 350788-350789Kevin P. Fleming
........ r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites are properly installed on Debian-style distributions. * Don't specify a specific version of libgmime; newer versions are available now and acceptable. * Install libsrtp so that res_srtp can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines Correct some 'set-but-not-used' variable warnings. ........ Merged revisions 350788-350789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350790 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Run bootstrap.sh for the for the ASTERISK-18929 fixKinsey Moore
configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350737 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Correct eventtype names in cel_odbc and cel_pgsql sample filesRichard Mudgett
........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Make sure asterisk builds on OpenBSDKinsey Moore
OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350731 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Set port to a default sane value if a bogus one is provided when parsing ↵Mark Michelson
hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350680 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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