asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	e1d3fc357c	Update documentation for CHANNEL function Document that you can read/write the 'accountcode' and 'amaflags' on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	0442f6fa95	Fixed spurious rebuilds of func_version. func_version.so was being rebuilt every time, because build.h was changing every build, because of the cleantest dependency that was added in r384410 to fix parallel make bugs. Now build.h will only be created if it does not exist, which was the original behavior of the Makefile. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	03b6927091	Fix parallel make problems. Occasionally, make -j would fail due to missing includes, or other unusual errors. This was due to the 'cleantest' target, which was designed to force a make clean when some change in the code would cause the typical depedency checking to fail. Several targets in the main Makefile did not depend upon cleantest, hence would run in parallel to it. By adding the dependency, make -j runs happily now. Review: https://reviewboard.asterisk.org/r/2418/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	86f663d7f4	app_voicemail: Add blank argument to externnotify if no context argument At least one call to run_externnotify provides a NULL context parameter and because the snprintf statement doesn't account for a NULL context parameter, it simply writes '(null)' to the arguments string instead. This patch makes it write two quotes back to back for that argument instead in the event of a NULL context. (closes issue ASTERISK-18207) Reported by: Barry L. Kline Patches: modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	1bf3009218	Address uninitialized conditional that valgrind found git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	85456f605e	Fix a file descriptor leak in off nominal path While looking at the security vulnerability in ASTERISK-20967, Walter noticed a file descriptor leak and some other issues in off nominal code paths. This patch corrects them. Note that this patch is not related to the vulnerability in ASTERISK-20967, but the patch was placed on that issue. (closes issue ASTERISK-20967) Reported by: wdoekes patches: issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	36706e0a93	Fix white noise on SRTP decryption When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c2a168d17c	AST-2013-003: Prevent username disclosure in SIP channel driver When authenticating a SIP request with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The information is disclosed when: * A "407 Proxy Authentication Required" response is sent instead of a "401 Unauthorized" response * The presence or absence of additional tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response is sent instead of "403 Forbidden" response after a retransmission * Retransmission are sent when a matching peer did not exist, but not when a matching peer did exist. This patch resolves these various vectors by ensuring that the responses sent in all scenarios is the same, regardless of the presence of a matching peer. This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of the testing and the solution to this problem was done by Walter as well - a huge thanks to his tireless efforts in finding all the ways in which this setting didn't work, providing automated tests, and working with Kinsey on getting this fixed. (closes issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	88784bb120	AST-2013-002: Prevent denial of service in HTTP server AST-2012-014, fixed in January of this year, contained a fix for Asterisk's HTTP server for a remotely-triggered crash. While the fix put in place fixed the possibility for the crash to be triggered, a denial of service vector still exists with that solution if an attacker sends one or more HTTP POST requests with very large Content-Length values. This patch resolves this by capping the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with Content-Length greater than this cap will not result in any memory allocation. The POST will be responded to with an HTTP 413 "Request Entity Too Large" response. This issue was reported by Christoph Hebeisen of TELUS Security Labs (closes issue ASTERISK-20967) Reported by: Christoph Hebeisen patches: AST-2013-002-1.8.diff uploaded by mmichelson (License 5049) AST-2013-002-10.diff uploaded by mmichelson (License 5049) AST-2013-002-11.diff uploaded by mmichelson (License 5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7079459be4	Resolve deadlock between SIP registration and channel based functions In r373424, several reentrancy problems in chan_sip were addressed. As a result, the SIP channel driver is now properly locking the channel driver private information in certain operations that it wasn't previously. This exposed two latent problems either in register_verify or by functions called by register_verify. This includes: * Holding the private lock while calling sip_send_mwi_to_peer. This can create a new sip_pvt via sip_alloc, which will obtain the channel container lock. This is a locking inversion, as any channel related lock must be obtained prior to obtaining the SIP channel technology private lock. * Holding the privat elock while calling sip_poke_peer. In the same vein as sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing the same locking inversion. Note that this locking inversion typically occured when CLI commands were run while a SIP REGISTER request was being processed, as many CLI commands (such as 'sip show channels', 'core show channels', etc.) have to obtain the channel container lock. (issue ASTERISK-21068) Reported by: Nicolas Bouliane (issue ASTERISK-20550) Reported by: David Brillert (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue ASTERISK-21296) Reported by: Gabriel Birke git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	21171e1684	Resolve deadlock between pending CDR and batch CDR locks r375757 attempted to resolve a race condition between multiple submissions of CDRs while in batch mode from attempting to destroy the scheduled batch submission by extending the batch CDR lock. Unfortunately, this causes a deadlock between the pending CDR lock and the batch CDR lock. This patch resolves the intent of r375757 by simply providing a new lock that protects the scheduling of the batches. The original batch CDR lock is kept to protect manipulation of the batch CDR settings, but has been placed such that it is not held when the pending lock is held. Thanks to Chase Venters for providing lock analysis on the issue. (issue ASTERISK-21162) Reported by: Chase Venters git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	37b7df75d4	Fix multi-station answer race condition. When an SLA trunk is ringing (inbound call on the trunk) Asterisk will make outbound calls to the stations that have that trunk. If more than one station answers the call at the same time, all channels other than the first one to answer are left in a bad state. The channel gets leaked, is not connected to anything, and there's no way to get rid of it. We now properly clean up these losing channels by hanging up on them. Since they lost the race, as we process their answer, there is no ringing trunk for them to answer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4f1021ad64	Set the CALLERID(dnid-num-plan) for incoming ISDN calls. The CALLEDTON channel variable is set for incoming ISDN calls to the lower 7 bits of the Q.931 type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan) should have the same value. (closes issue ASTERISK-21248) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	8c4732636f	Properly delimit post data in res_config_curl. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	6bd4feca22	Have func_curl log a warning when a curl request fails. Review: https://reviewboard.asterisk.org/r/2403/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	5cf67a7ab2	Removed codecs/g722/*.i on make clean git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	fd4b11284e	tcptls: Prevent unsupported options from being set AMI, HTTP, and chan_sip all support TLS in some way, but none of them support all the options that Asterisk's TLS core is capable of interpreting. This prevents consumers of the TLS/SSL layer from setting TLS/SSL options that they do not support. This also gets tlsverifyclient closer to a working state by requesting the client certificate when tlsverifyclient is set. Currently, there is no consumer of main/tcptls.c in Asterisk that supports this feature and so it can not be properly tested. Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d0306b2031	When a session timer expires during a T.38 call, re-invite with correct SDP When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b3f149a880	Fix processing of call files when using KQueue on OS X In certain situations, call files are not processed when using KQueue with pbx_spool. Asterisk was sending an invalid timeout value when the spool directory is empty, causing the call to kevent to error immediately. This can create a tight loop, increasing the CPU load on the system. (closes issue ASTERISK-21176) Reported by: Carlton O'Riley patches: kqueue_osx.patch uploaded by coriley (License 6473) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	320b4e8bda	Fix whitespace in AST_EXT_LIB_CHECK macro. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	7d7e11de51	Fix Sorting Order For Parking Lots Stored In Static Realtime When retrieving the parking lots from a MySQL database table, the current order is "filename, cat_metric desc, var_metric asc, category". If there are multiple parking lots with the same cat_metric but different categories, everything is being sorted on cat_metric first resulting in errors when loading the parking lots. This patch fixes the problem by sorting on the category field first, then the cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	85f0aadb86	Update Contributed Realtime Schema Files - IPv6 Addresses This commit updates some fields in the contributed realtime schema files to handle IPv6 addresses. (closes issue ASTERISK-21173) Reported by: Torrey Searle Patches: realtime_sql.patch Torrey Searle (license 5334) asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3f78df26b5	Include the Username field in SIP Registry events when Status is registered In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	143ebd5665	Correct app_page documentation The 'A' and 'n' options for Page() mention that the announcement will be played simultaneously. This is not necessarily the case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	6008772ca3	Fix several unreleased mutex locks that cause problem with processing calls (Closes issue ASTERISK-21119) Reported by: Daniel Bohling Tested by: Daniel Bohling git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	45b88e8f04	Prevent deadlock in chan_iax2 when attempting to set caller ID A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it already holds the iax2 private lock and improperly fails to obtain the channel lock before calling ast_set_callerid. By not safely obtaining the channel lock, a locking inversion can take place, causing a deadlock. This patch solves this by calling the required deadlock avoidance functions that obtain the channel lock before setting the caller ID. Thanks to Pavel for fixing my syntax errors and testing this patch out. (closes issue ASTERISK-21128) Reported by: Pavel Troller Tested by: Pavel Troller patches: ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283) ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fc5e76b96a	Let channels joining a MeetMe conference opt out of the denoiser For some channel drivers, specifically those that have a varying rate in the number of audio samples, the audio quality for a MeetMe conference can be exceedingly poor. This is due to a unilateral application of the DENOISE function in func_speex to channels joining the conference. The denoiser function in the speex library is initialized with the number of audio samples in each sample that will be provided to it. If the number of audio samples changes, the denoiser has to be thrown away and re-initialized. While this could be worked around by removing func_speex, that doesn't help if you actually use the denoiser with other channels on the system. This patches does the following: * Checks for the presence of func_speex as opposed to codec_speex when determining if the DENOISE function is present (which is where the function is actually implemented) * Adds an option to MeetMe 'n' that causes the denoiser to not be applied to a channel when it joins. This keeps the current behavior the default, but let's users disable the denoiser if it causes problems on their system. Review: https://reviewboard.asterisk.org/r/2358 (closes issue AST-1062) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	011b214822	Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog is forked. (closes issue ASTERISK-20638) Reported by: eelcob Patches: pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	54ce2c1491	Fix the configure script over here as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	a6a3a8198a	Consider linux-gnuspe as linux-gnu * The powerpcspe Linux port uses linux-gnuspe as the OS string. * Our build system shouldn't really care for that, so just call it linux-gnu. * Original report: Roland Stigge , http://bugs.debian.org/701505 Review: https://reviewboard.asterisk.org/r/2357/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	68d2599d39	Correct RPID parsing for unquoted display-name. Parsing Remote-Party-ID will now succeed if display-name is of the *(token LWS) kind and not just the quoted-string kind. Review: https://reviewboard.asterisk.org/r/2341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	ee4d92bb6b	Remove unneeded linux-gnueabi* As of r380520 the configure scripts converts the value of linux-gnueabi* of OSARCH to "linux-gnu". So no point in testing for those values. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b028e09f9e	Clean up use of va_end/va_args in res_config_mysql There were several problems using variadic argument macros in res_config_mysql. * Improper use of va_end. Multiple calls to va_end were possible resulting in an unbalanced matching of va_start/va_end. * Calls to va_arg after a possible encounter of a SENTINEL value. This patch corrects those errors. (closes issue ASTERISK-19451) Reported by: wdoekes patches: ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	37f8a4a299	Set the sin_family on the bind address socket during initialization Somehow, chan_jingle has managed to operate for years without setting the sin_family on its bindaddr socket. This patch properly sets the field during initial module load to AF_INET. Note that the patch on the issue was modified slightly to change the initialization of the socket from allocation of a chan_jingle private to the module initialization, as the bindaddr object (which is static) only needs to have the address set once. (closes issue ASTERISK-19341) Reported by: andre valentin patches: 0105-chan_jingle.patch uploaded by avalentin (License 6064) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	bd9742163d	Don't display the AMI ALL class authorization for users if they don't have it When converting AMI class authorizations to a string representation, the method always appends the ALL class authorization. This is especially important for events, as they should always communicate that class authorization - even if the event itself does not specify ALL as a class authorization for itself. (Events have always assumed that the ALL class authorization is implied when they are raised) Unfortunately, this did mean that specifying a user with restricted class authorizations would show up in the 'manager show user' CLI command as having the ALL class authorization. Rather then modifying the existing string manipulation function, this patch adds a function that will only return a string if the field being compared explicitly matches class authorization field it is being compared against. This prevents ALL from being returned unless it is actually specified for the user. (closes issue ASTERISK-20397) Reported by: Johan Wilfer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	726b60ef30	Make ParkAndAnnounce return to priority + 1 when return context is not defined The ParkAndAnnounce application documentation for the optional return_context parameter states the following: return_context The goto-style label to jump the call back into after timeout. Default 'priority+1'. Unfortunately, the application was sending the channel back into the dialplan at 'priority', which is the ParkAndAnnounce application call. This causes an infinite loop of the channel constantly being parked, announced, timed out, parked, announced, timed out... while fun, especially for those callers you wish to drive to the end of madness, this was not the intent of the application. (closes issue ASTERISK-20113) Reported by: serginuez patches: app_parkandannounce.diff uploaded by serginuez (License 6405) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6d295bcabf	Properly detect launchd Asterisk was a little too pro-active in claiming that it found launchd. On systems without launchd - such as FreeBSD - this resulted in certain items in Asterisk that conflict with launchd to not be selectable, such as res_timing_kqueue. (closes issue ASTERISK-20749) Reported by: Oleg Baranov git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	5143f6f85c	Write the correct callid to the data1 field in queue_log for transfer events. The incorrect callid was being written to the "data1" field in queue_log table for transfer events. The callid of the queue was being written instead of the transfer target's callid. This now gets the correct "transfer to" number and places that in the "data1" field of the queue_log table when a transfer event is triggered. (closes issue ASTERISK-19960) Reported by: vladimir shmagin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	706235d7ee	Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample The "registertrying" option was removed in r343220. The "rtp_engine" option was added in r186078 but erroneously named "engine" in the sample. Note that there is no global sip setting for a different engine. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4b674040e2	End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO because it isn't a real hangup. It doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside of a bridge. (issue ASTERISK-20492) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	775e80226e	Don't throw a spurious error when using DBdeltree The function call ast_db_deltree returns the number of row deleted, or a negative number if it failed. DBdeltree was treating any non-zero return as an error, causing a spurious verbose error message to be displayed. This patch handles the return code of ast_db_deltree correctly. (closes issue ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff uploaded by ianc (License #5955) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5fea5fd630	Do not allow native RTP bridging if packetization of media streams differs. The RTP engine will no longer allow for local and remote native RTP bridges if packetization of streams differs. Allowing native bridging in this scenario has been known to cause FAX failures. (closes ASTERISK-20650) Reported by: Maciej Krajewski Patches: ASTERISK-20659.patch uploaded by Mark Michelson (License #5049) Review: https://reviewboard.asterisk.org/r/2319 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	42b6f880f4	Properly load say.conf upon reload of module app_playback. If say.conf did not exists prior to originally loading module app_playback it would not load on subsequent reloads of the module once it had been created. This occurred because upon reload of the app_playback module it would only load a new configuration if an old one had previously existed. This fix simply removed the association between checking if an old configuration existed and the loading of the new one. (closes issue ASTERISK-20800) Reported by: pgoergler git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	6c76b18b85	Fixed failing test from r380696. When I added my extensive suite of session timer unit tests, apparently one of them was failing and I never noticed. If neither Min-SE nor Session-Expires is set in the header, it was responding with a Session-Expires of the global maxmimum instead of the configured max for the endpoint. (issue ASTERISK-20787) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8aa99bc3f8	Separate option_types[] from the struct definition. Updated the option_types[] doxygen comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9ff89704b6	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	165410c002	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fe20f6bb6b	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c1ad425b5f	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e365562f91	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. (closes issue ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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