asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	ef0433e6bc	app_confbridge: Fix error messages on exiting conference. A marked user ending a conference with only end_marked users generates error messages: ERROR[0000][C-00000000]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' * The MULTI_MARKED state was doing too much when it was kicking out the end_marked users from the conference. The kicked out users will clean up after themselves when they exit the conference. (closes issue ASTERISK-20991) Reported by: Jeremy Kister Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6754f20f59	app_page: Fixup application XML documentation typos and inaccuracies. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	186f8f4a39	Because the compiler can check types with a struct copy and memcpy() cannot. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	102c20519e	Separate option_types[] from the struct definition. Updated the option_types[] doxygen comment. ........ Merged revisions 380853 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	e7e730f973	Fix how we build pjproject. Allow parallel builds, better tolerate failures, build faster. This also stops running dependencies before top-level configure has been run. (closes issue ASTERISK-20815) Review: https://reviewboard.asterisk.org/r/2292/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	d9d5028b01	Ignore warnings caused by PJ_TODO()s in pjproject. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	47f8394517	Fix a few compiler warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	1412adf576	Process session timers, even if Session-Expires header is missing Previously, Asterisk only processed session timer information if both the 'Supported: timer' and 'Session-Expires' headers were present. However, the Session-Expires header is optional. If we were to receive a request with a Min-SE greater than our configured session-expires, we would respond with a 'Session-Expires' header that was too small. This patch cleans the situation up a bit, always processing timer information if the 'Supported: timer' header is present. (closes issue ASTERISK-20787) Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/ ........ Merged revisions 380696 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	d865240168	Add support for parallel builds of pjproject. Also adds proper dependency checking, and direct .a file targets. We don't take advantage of this currently, but we will soon. (issue ASTERISK-20815) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	ff0d016390	Always check for libm, regardless of configure options. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	e02189ffa1	Remove a cross-compile workaround. ar and ranlib can be easily detected with autoconf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2d1b7d4732	Make CHECK_BLOCKING() debug message more useful. Change the displayed pthread value to hex format so it can be easily matched with CLI core show threads or gdb. ........ Merged revisions 380611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	686f2c50c7	chan_dahdi: Fix "dahdi show channels group" for groups greater than 31. The variable type used was not large enough to hold a group bit field. ........ Merged revisions 380572 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	49959b511f	Support building Asterisk for Raspberry Pi/Raspbian with hard-float support Building Asterisk on Raspbian with hard-float support fails as it uses the string 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'. This patch modifies the configure script for Asterisk such that it will match on any string beginning with 'linux-gnueabi', as opposed to requiring an explicit match. (closes issue ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse (license 6459) linux-gnueabihf-autoconf.patch uploaded by Christian Hesse (license 6459) ........ Merged revisions 380520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	df83528506	Unregister SIP provider API if module load is declined A user in #asterisk ran into a problem where a configuration error prevented the chan_sip module from being loaded. Upon fixing their configuratione error, they could no longer load the chan_sip module. This was because the configuration checking happened after the SIP provider was registered with the Asterisk core, and subsequent attempts to load the SIP module failed as the provider was already registered. Since we want to detect any failure in registering chan_sip as early as possible (as that could be emblematic of a deeper mismatch between module and Asterisk core), this patch does not change the registration location, but does ensure that if a module load is declined, we unregister the module as the SIP api provider. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	896bf5e5a8	Perform case insensitive comparisons for T.38 attributes RFC5347 section 2.5.2 states the following: ... The attribute "T38MaxBitRate" was once incorrectly registered with IANA as "T38maxBitRate" (lower-case "m"). In accordance with T.38 examples and common implementation practice, the form "T38MaxBitRate" SHOULD be generated by implementations conforming to this package. In general, it is RECOMMENDED that implementations of this package accept lowercase, uppercase, and mixed upper/lowercase encodings of all the T.38 attributes. ... Asterisk currently does not perform case insensitive matching on the T.38 attributes. This causes the T38MaxBitRate attribute to be negotiated at 2400 baud instead of 14400 (or whatever value you actually wanted). This patch makes it so that when we compare T.38 attributes, we do so in a case insensitive fashion. Note that while the issue reporter did not directly write the patch, they contributed to it (and would have provided one themselves if the license had gone through a tad faster), and hence get attribution for it. Review: https://reviewboard.asterisk.org/r/2298/ (closes issue ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill ........ Merged revisions 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2a458061b1	Fix memory leak in res_calendar_icalendar The ICalendar module had a systemic memory leak on each fetch of data from the ICalendar source. The previous fetched data was not being properly disposed. This patch makes it so that before each fetch of data, we dispose of the previously fetched data. (closes issue ASTERISK-21012) Reported by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7b41f52cf2	chan_agent: Prevent multiple channels from logging in as the same agent. Multiple channels logging in as the same agent can result in dead channels waiting for a condition signal that will never come because another channel thread stole it. A symptom is chan_sip repeatedly generating warning messages about rescheduling autodestruction of dialogs with an agent channel owner. * Made only login_exec() (the app AgentLogin) clear the agent_pvt->chan pointer to prevent multiple channels from logging in as the same agent. agent_read(), agent_call(), and agent_set_base_channel() no longer disconnect the agent channel from the agent_pvt. This also eliminates the need to keep checking for agent_pvt->chan being NULL. * Made agent_hangup() not wake up the AgentLogin agent thread until it is done. * Made agent_request() not able to get the agent until he has logged in and any wrapup time has expired. * Made agent_request() use ast_hangup() instead of agent_hangup() to correctly dispose of a channel. * Removed agent_set_base_channel(). Nobody calls it and it is a bad thing in general. * Made only agent_devicestate() determine the current device state of an agent. Note: Agent group device states have never been supported. Review: https://reviewboard.asterisk.org/r/2260/ ........ Merged revisions 380364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	765c1ac026	Corrected crypto tag in SDP ANSWER for SRTP. (again) The original fix (r380043) for getting Asterisk to respond with the correct tag overlooked some corner cases, and the fact that the same code is in 1.8. This patch moves the building of the crypto line out of sdp_crypto_process(). Instead, it merely copies the accepted tag. The call to sdp_crypto_offer() will build the crypto line in all cases now, using a tag of "1" in the case of sending offers. (closes issue ASTERISK-20849) Reported by: José Luis Millán Review: https://reviewboard.asterisk.org/r/2295/ ........ Merged revisions 380347 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	db03ae6c9a	call_parking: Make sure fallbacks are used when lacking a flat channel exten A regression was introduced which removed automatic fallback behavior from the PBX. This behavior was used by call parking (or at least documented as how the feature works) in order to select an extension when the flat channel extension wasn't available from the comebackcontext. Parking now handles the fallbacks internally in order to keep behavior matching with how it is documented. (closes issue ASTERISK-20716) Reported by: Chris Gentle Review: https://reviewboard.asterisk.org/r/2296/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3a6bc03c57	Ensure that a declined media stream is terminated with a '\r\n' In r369028, chan_sip's processing of media streams in an SDP was modified to better handle multiple offered media streams. Part of that change modified how streams were declined. Previously, declined media streams were not handled in an RFC compliant manner; now, we set the port number to 0 in the media stream definition and proceed on with the next media stream. Unfortunately, the formatting of the declined media stream forgot to append a '\r\n' to the end of the media stream. This is normally added to the accepted media streams later on in the processing of the SDP. Since the declined media stream uses a different buffer than the accepted media streams (and is a malloc'd buffer as opposed to a struct ast_str), it's easier to just slap the '\r\n' on the declined media stream buffer rather than attempt to append it later on. So, that's what we do. And now some devices (and probably some providers) will be a bit happier (but probably not terribly happy, since we just rejected something they offered). Review: https://reviewboard.asterisk.org/r/2297/ (closes issue ASTERISK-20908) Reported by: Dennis DeDonatis Tested by: Dennis DeDonatis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3f49b5faba	Update configure script to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the configure script fails due to grep returning multiple matches for the pattern it searches for. This patch updates the pattern matching to return only the actual version for the symbol searched for, PTLIB_VERSION. (closes issue ASTERISK-20980) Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch uploaded by Stefan Reuter (license 5339) ........ Merged revisions 380297 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c835ff7cd4	Correct the number of available call numbers in IAX2. There is currently an edge case where call number 32768 might be allocated for a call, even though the IAX2 protocol requires call numbers be only 15 bits. This resulted in some unpredictable behavior when call number 32678 is chosen. This patch was mostly written by Richard Mudgett via ReviewBoard. I'm just committing it. Review: https://reviewboard.asterisk.org/r/2293/ ........ Merged revisions 380254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	ae5c07f0f1	Change cleanup ordering in filestream destructor. This patch came about due to a problem observed where wav files had an empty header. The header is supposed to be updated in wav_close(). It turns out that this was broken when the cache_record_files option from asterisk.conf was enabled. The cleanup code was moving the file to its final destination before running the close() method of the file destructor, so the header didn't get updated. Another problem here is that the move was being done before actually closing the FILE *. Finally, the last bug fixed here is that I noticed that wav_close() checks for stream->filename to be non-NULL. In the previous cleanup order, it's checking a pointer to freed memory. This doesn't actually cause anything to break, but it's treading on dangerous waters. Now the free() of stream->filename is happening after the format module's close() method gets called, so it's safer. Review: https://reviewboard.asterisk.org/r/2286/ ........ Merged revisions 380210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	dc1100b899	Fix Some Configured Conference Bridge Sounds Not Being Set The "sound_only_one" sound was not being set even though it was configured. In looking into this, I found that the "join" and "leave" prompts were not being set either. (closes issue ASTERISK-20898) Reported by: Stephan Tested by: Stephan Patches: asterisk-20898-custom-sounds-ignored.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2289/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	7be62f9931	Corrected crypto tag in SDP ANSWER for SRTP. When Asterisk responds with an SDP ANSWER for SRTP, it had the code to correctly fill in the crypto data, which was overwritten by a call to sdp_crypto_offer. Corrected the situation by changing sdp_crypto_offer to not replacing crypto data if it already exists. (closes issue ASTERISK-20849) Reported by: José Luis Millán Tested by: Iñaki Baz Castillo Patches: fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	322525bece	Correct documentation for ConfbridgeList AMI action The documentation for ConfbridgeList states that the Conference field is optional. That's not really the case: if you fail to provide a Conference number, the command will kick back an error. (closes issue AST-1090) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	24b4a30a09	Attempt to be more helpful when using a bad ao2 object pointer. Put the external obj pointer in the message instead of the internal version. ........ Merged revisions 379963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	c7d22a6af6	res_fax_spandsp: fix t38 transmission bug caused by not returning success This patch fixes the problem, but the issue includes a test which is still being considered for the automated test suite. (issue ASTERISK-20919) Reported by: NITESH BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	36d3124745	app_meetme: Use new prompts for administrator menu The old prompts for the administrator menu were inadequate. They didn't mention that the menu had additional options through the 8 key and pressing the 8 key wouldn't reveal what those options were. This patch fixes all of that while also organizing code pertaining to each individual menu type which was previously all stored in one gigantic function along with many of the basic conference functions. (closes issue AST-996) Reported by: John Bigelow Review: http://reviewboard.digium.internal/r/360/ ........ Merged revisions 379885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	9faedde313	Fix station ringback; trunk hangup issues in SLA This patch fixes two bugs: * If an outbound call is made from a SLA phone using SLAStation, then there is no ringtone audible to the phone that originates the call. The indication of the ringing was not being passed to the SLA station; this patch fixes that by passing through the progress indications. * If an SLA station hangs up before the called party answers, then the channel to the called party continues to ring until a timeout occurs. If the called party manages to answer, Asterisk attempts to connect the called party to a non-existant MeetMe room. This patch corrects the behavior by abandoning the call attempt if it detects that the SLA station is no longer in use while attempting to call the called party. Review: https://reviewboard.asterisk.org/r/2275/ (closes issue ASTERISK-20462) Reported by: dkerr patches: asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches: asterisk-11-bugid20440.patch uploaded by dkerr (license 5558) asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558) ........ Merged revisions 379825 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	df6a5fb23f	confbridge: Minor fixes playing user counts to the conference. * Generate a warning message if sound files do not exist when trying to play the user count to the conference. Use the new helper routine sound_file_exists() for consistency. * Put the new user into autoservice when playing user counts to the conference. * Check the return value of ast_bridge_impart(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e7d0d2bd4c	Update init.d scripts to handle stderr; readd splash screen for remote consoles When r376428 was commited to re-order start up sequences to be more tolerant of forking with thread primitives, a few items were changed that caused changes in behavior on some distros. This includes: * Not displaying the splash screen on a remote console. * Displaying an error message on stderr when a remote console cannot connect to a running instance of Asterisk. In the first case, the splash screen was re-added (thanks to Michael L. Young). In the second case, the various init.d scripts were modified to pipe stderr to /dev/null, as the error message is useful - if you execute a remote console or a remote console command execution and it fail, it should tell you. Note that the error message was always present, it just failed to be printed prior to r376428. Much thanks to the folks who quickly reported this problem, provided solutions, and promptly tested the various init.d scripts on a variety of distros. (closes issue ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches: asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283) ........ Merged revisions 379760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379777 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	dbb4b6c8b6	Prevent segfault for interpolated iLBC frames When iLBC is being used with a jitter buffer and the jb has to interpolate frames, it generates frames with a null pointer and a non-zero datalen. This is now handled properly. (closes issue ASTERISK-20914) Reported By: John McEleney Patches: ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 379718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	3db7711a63	Fix device call logging issues in skinny Skinny device call logging (ie missed, place and received calls) has issues because the incorrect sequence of callstates is/can be sent to the device. This patch removes some extra callstate updates driven by forces external to skinny and ensures the needed intermediary callstate messages are sent. (closes issue ASTERISK-20964) Reported by: wedhorn Tested by: snuffy, myself Patches: ast11-skinny-calllog01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	bd958b3250	Add LDAP libraries to install script Add LDAP dev package to Debian/Ubuntu install list. Existed in Redhat already. (issue ASTERISK-20886) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	89949f1c5f	Fix crash in app_minivm when mime encoding string An incorrect string initializations was left in ast_str_encode_mime from the patch that converted string manipulations to use ast_str strings (r191140). The string initialization causes a crash when ast_str_set is called on the string later on in the function. (closes issue ASTERISK-18697) Reported by: Chris Boot patches: minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr Tested by: Chris Warr ........ Merged revisions 379608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	cb5f7233b9	Fix issues with skinny sessions Fixes a couple of issues with the way skinny handles sessions by ensuring sessions aren't used after being freed. Some other minor changes. Review: https://reviewboard.asterisk.org/r/2272/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	bb450bafd7	Add builtin roundf() for systems lacking it. (closes issue ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276 Reported-by: Ovidiu Sas ........ Merged revisions 379547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f08028c445	Fix astcanary startup problem due to wrong pid value from before daemon call When Asterisk forks itself into the background via a call to daemon, it must re-set the pid value of the new process. Otherwise, astcanary gets the pid value of the process before the fork, which prevents it from running. Asterisk eventually starts lowering its priority, as it can no longer communicate with the proverbial canary in the coal mine. This patch ensures that the correct process identifier is used by astcanary. Note that this is getting committed to 10 as a regression fix. (closes issue ASTERISK-20947) Reported by: Jakob Hirsch Tested by: mjordan patches: asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113) ........ Merged revisions 379509 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379510 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c32dc894ca	Fix regression in Confbridge user count When the restructuring work got committed to Confbridge in r375470 to fix many open issues, it caused a regression in the reported count of users when conference information was requested via CLI or manager. This corrects the user count and user information displayed when listing conference information from the CLI and manager. (closes issue ASTERISK-20938) Reported By: Timo Teras Patches: confbridge-list.patch uploaded by Timo Teras (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	af6b4fed4f	Specify the -rpath linker flag when prefix != /usr. This allows Asterisk to start without having to specify the LD_LIBRARY_PATH. This can be disabled by passing --disable-rpath to configure. (closes issue ASTERISK-20407) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/2132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1752c6279d	app_voicemail: Improve msg_id handling app_voicemail will no longer issue error messages when it retrieves an msg_id with a NULL value from realtime and will instead simply populate the msg_id field with a newly generated msg_id. In addition, this patch changes the way msg_ids are generated to eliminate certain causes of duplicate IDs appearing within a single system. In addition, when messages are copied, they will now receive a new msg_id. (closes issue ASTERISK-20717) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/2220/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	f04706fde4	Fix Record-Route parsing for large headers. Record-Route parsing copied the header into a char[256] array, which can be a problem if the header is longer than that. This patch parses the header in place, without the copy, avoiding the issue. In addition to the original patch, I added a unit test for the new get_in_brackets_const function. (closes issue ASTERISK-20837) Reported by: Corey Farrell Patches: chan_sip-build_route-optimized-rev1.patch uploaded by Corey Farrell (license 5909) (with minor changes by dlee) ........ Merged revisions 379392 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	b80431fe60	Fix issue where chan_mobile fails to bind to first available port Per the bluez API, in order to bind to the first available port, the rc_channel field of the socket addressing structure used to bind the socket should be set to 0. Previously, Asterisk had set the rc_channel field set to 1, causing it to connect to whatever happens to be on port 1. We could probably not explicitly set rc_channel to 0 since we memset the struct earlier, but explicitly setting it will hopefully prevent someone from coming in and setting it to some explicit port in the future. (closes issue ASTERISK-16357) Reported by: challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin, eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by Nikolay Ilduganov (license 6253) ........ Merged revisions 379342 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	97addb10b3	Further fix misinformation in the description of manager MailboxStatus command. The description still claimed that it returned the number of messages rather than whether there were messages waiting. ........ Merged revisions 379310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	d42db66d4d	Reduce number of packages install_prereq installs on Debian systems. 'search' will look for any package containing the name provided, so we need to force a more exact search. ........ Merged revisions 379276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cb65f124c5	Reduce call-id logging resource usage. Since there is no need for the call-id logging ao2 object to have a lock, don't create it with one. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	64843422b7	chan_misdn: Fix compile error. (issue ASTERISK-15456) ........ Merged revisions 379226 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	fb72636fd8	Let documentation reference links specify which module they're linking to Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref links have to specify which reference they're referring to. The various documentation parsers can interpret the module attribute however they want in order to construct the appropriate links. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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