asterisk

Commit Graph

Author	SHA1	Message	Date
Alexandr Anikin	075b8385a0	Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) ........ Merged revisions 352807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352817 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f4d98aeb28	Copy amaflags to sip_pvt from peer during create_addr_from_peer For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner ........ Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352756 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	ed32b1c098	Merged revisions 352705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352705 \| alecdavis \| 2012-01-26 19:33:11 +1300 (Thu, 26 Jan 2012) \| 27 lines Merged revisions 352704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352704 \| alecdavis \| 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) \| 20 lines Cleanup dialog-info+xml Notify dialog Make similar to other Notify messages. sample output: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="715" state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523"> <state>terminated</state> </dialog> </dialog-info> Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5be89b07e2	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	9ee8a74461	Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". A long time ago, in a land far far away, we added "asterisk/ast_version.h", which provides the ast_get_version() and ast_get_version_num() functions. These were added so that modules that needed the version information for the Asterisk instance they were loaded in could actually get it (as opposed the version that they were compiled against). We changed everything in the tree to use the new mechanism (although later main/test.c was added using the old method). However, the old mechanism was never removed, and as a result, new code is still trying to use it. This commit removes asterisk/version.h and replaces it with a header that will generate a compile-time error if you try to use it (the error message tells you which header you should use instead). It also removes the Makefile and build_tools bits that generated the file, and it updates main/test.c to use the 'proper' method of getting the Asterisk version information. This is an API change and thus is being committed for trunk only, but it's a fairly minor one and definitely improves the situation for out-of-tree modules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	261cde4675	Blocked revisions 352616 ........ Avoid unnecessary rebuilds of main/test.c. main/test.c includes "asterisk/version.h", when it should include "asterisk/ast_version.h" instead (and it should use the ast_get_version() and ast_get_version_num() functions). This commit modifies it to extract the Asterisk version information using the proper APIs, and as a result means that main/test.c no longer needs to be rebuilt when a Subversion checkout is updated or modified. ........ Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	080ea28515	Remove some extraneous debugging from registry memleak fix ........ Merged revisions 352551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352556 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	4bf5e3716e	Clean up some SIP registry-related memory leaks 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ ........ Merged revisions 352514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352515 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	50de9578aa	Eliminate unnecessary rebuilds of main/format*.c. These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. ........ Merged revisions 352516 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	973aeabf2d	Redocuments sip types peer, user, friend in sip.conf.sample There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352512 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	213f7a65b5	Fix channel opaquification of stringfields for chan_vpb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	0fe9043233	Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. (closes issue ASTERISK-16550) reported by: Olle Johansson ........ Merged revisions 352424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352430 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b68b7e55ac	Set core sounds version to 1.4.22. 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	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields 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	14 years ago
Richard Mudgett	2144ba5df2	Fix locking issues with channel datastores in func_odbc.c. * 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	14 years ago
Joshua Colp	84aea92ba7	Blocked revisions 352288 ........ 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	14 years ago
Mark Michelson	c3c6b5a0ba	Fix grammar of comment. ........ 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	14 years ago
Mark Michelson	0920c50341	Fix blind transfers from failing if an 'h' extension is present. 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	14 years ago
Matthew Jordan	59a42de303	Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer 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	14 years ago
Jonathan Rose	de09749470	Add an announcement option to music-on-hold - plays sound when put on 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	14 years ago
Jonathan Rose	a1bef6041d	Adds the ability to stop specific mixmonitors by using unique IDs set at 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	14 years ago
Richard Mudgett	74508e3bca	Fix sip_cfg.notifycid to be set with the defined enum values. 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	14 years ago
Richard Mudgett	20c6ff71b6	Fix ast_app_dtget() time unit inconsistency. 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	14 years ago
Mark Michelson	02408a2476	Remove XXX comment that is not necessary. ........ 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	14 years ago
Mark Michelson	ab8ba431b6	Fix RTP reference leak. 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	14 years ago
Richard Mudgett	d0c765497d	Make CLI sip show channel list the complete route set. (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	14 years ago
Kinsey Moore	c6fd4f5d74	SIP session timeout AMI event 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	14 years ago
Mark Michelson	778fa4abaf	Various parking improvements. * 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	14 years ago
Mark Michelson	b98a25ef93	Prevent potential buffer overflow on AMI MixMonitor command. 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	14 years ago
Kinsey Moore	43621b05a9	More corrections for the ilbc code 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	14 years ago
Kinsey Moore	d61920b6d9	Restore LSF_check function calls from set/unused variable removal 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	14 years ago
Kinsey Moore	75243988d5	Remove more set, but unused variables in the ilbc codec 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	14 years ago
Jonathan Rose	1a6960099b	Adds setting of mwi_from field to check_auth_result check_peer_ok (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	14 years ago
Matthew Jordan	7a442b017c	Remove unused variable 'tmp' from helpfun in ilbc codec 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	14 years ago
Stefan Schmidt	f4f5ccf5d7	enable doxygen build for files in the channels/sip folder like reqresp_parser.c ........ 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	14 years ago
Richard Mudgett	ae32acfa3e	Misc minor fixes in reqresp_parser.c and chan_sip.c. * 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	14 years ago
Kinsey Moore	add6efc20c	Correct output of RTCP jitter statistics in SR and RR reports 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	14 years ago
Jonathan Rose	6fd0ac9dcd	Eliminates doubling the :port part of SIP Notify Message-Account headers. 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	14 years ago
Joshua Colp	ddf421bd5c	Prevent crash when an SDP offer is received with an encrypted video stream 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	14 years ago
Matthew Jordan	16adf6de8c	Include iLBC source code for distribution with Asterisk 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	14 years ago
Stefan Schmidt	f69fd136f4	The get_pai function in chan_sip.c didn't recognized a proper callerid name and 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	14 years ago
Walter Doekes	a9698d0241	Fix support for parallel building with make (-j). 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	14 years ago
Mark Michelson	f5dd17e558	Eliminate odd initialization of probation variable. ........ 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	14 years ago
Jonathan Rose	ee4cf38a27	Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. 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	14 years ago
Mark Michelson	5af788ccd0	Use built-in parsing functions for Contact and Record-Route headers. 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	14 years ago
Matthew Jordan	f86621eb93	Fix udptl issue with initial INVITE introduced by r351027 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	14 years ago
Russell Bryant	141dd18848	Merged revisions 351183 via svnmerge from 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	14 years ago
Terry Wilson	aacc158072	Ensure ACK retransmit & hangup on non-200 response to INVITE 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	14 years ago
Terry Wilson	fb5924ebe8	Don't prematurely stop SIP session timer 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	14 years ago

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