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Author	SHA1	Message	Date
Kinsey Moore	95ad8ea190	Close libsrtp properly ... Ensure that libsrtp is shutdown properly when res_srtp is unloaded. (closes issue ASTERISK-21719) Reported by: Corey Farrell Patches: res_srtp-library-shutdown.patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b7da14ecf4	Fix SendText AMI action to never return non-zero. ... AMI actions must never return non-zero unless they intend to close the AMI connection. (Which is almost never.) (closes issue ASTERISK-21779) Reported by: Paul Goldbaum git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	af1711d17e	Allow mISDN to send PROGRESS messsage. ... * Made isdn_msg_parser.c build a progress message with the mandatory progress indicator IE. (The mISDNuser NT state machine rejected sending the incomplete message.) Note: The associated mISDN and mISDNuser patches respectively are viewable here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200 http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes issue AST-1153) Reported by: Guenther Kelleter Patches: progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	777d526415	Add version.c to list of ignored files in the utils directory. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	6cbc641491	Fix memory leak in pbx_dundi ... pbx_dundi added an io context without removing it. This caused a memory leak when the module was unloaded. (closes ASTERISK-21718) Reported by Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by Corey Farrell (License #5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	276c5e6d45	Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit ... When we send out a CN packet (for instance, in the case of using rtpkeepalives), we are not setting the payload code properly. Also, we are setting the marker bit when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup(). 11 and trunk already use the appropriate function. * In 1.8, use ast_rtp_codecs_payload_code() * Remove the setting of the marker bit * Fix the debug message by incrementing the seqno after the debug message is set in order to display the correct seqno that was sent out (closes issue ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter Katzmann, Michael L. Young Patches: asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2500/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	0d9cc94441	chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail ... RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription The problem is that the State Notify requests rely on the 200OK reponse for pacing control and to not confuse the notify susbsystem. The issue is, the pendinginvite isn't cleared if a response isn't received, thus further notify's are never sent. The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure. (closes issue ASTERISK-21677) Reported by: Dan Martens Tested by: Dan Martens, David Brillert, alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2475/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	7124314fb7	Make SLA reload more paranoid. ... Reload support was originally not included for SLA. It was added later, but in a fairly non-traditional way. It basically sets a flag indicating that a reload is pending, and then waits for a time where it thinks everything SLA related is idle and unused, and *then* executes the reload. It does this because the reload process is destructive. It starts by throwing everything away and starting over. There are a number of problems with this approach. One of them is that the check to see if anything in use was incomplete. This patch makes it more complete and thus less likely for a crash to occur during reload processing. However, this approach still has problems so some much more significant reworking of this code will need to come in as a next step. Patch credit and testing by CoreDial, LLC. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cc950e6a80	Update utils Makefile to handle r387294 ... Alec's patch that added the Asterisk version to 'core show locks' angered the items in utils, as they exist somewhat outside of the Asterisk build system. Some day, this Makefile should get nuked from high orbit, but for now, include version.c in its list of stuff to pile in. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	05fb483da6	chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher ... RFC 4028 Section 10 if the side not performing refreshes does not receive a session refresh request before the session expiration, it SHOULD send a BYE to terminate the session, slightly before the session expiration. The minimum of 32 seconds and one third of the session interval is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the Session-Expires interval, or if the remote device was the refresher, asterisk would timeout at interval end. Now, when not refresher, timeout as per RFC noted above. (closes issue ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2488/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	d6b88f6627	chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher. ... RFC 4028 Section 7.2 "UACs MUST be prepared to receive a Session-Expires header field in a response, even if none were present in the request." What changed After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher. Symptom: After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device may respond with a much lower Session-Expires (180 in our case) value that it is now using. Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE. After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response. Fix: handle_response_invite() when 200OK, remove check for outbound and reinvite. (closes issue ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2463/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	00c14d8c0d	chan_dahdi: fix lower bound check with -ve integer conversion from a float ... Lower bound of a 16bit signed int is -32768 not -32767 (closes issue ASTERISK-21744) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	af6d77f108	Add Asterisk Version to core show locks ... Assist with reporting 'core show locks' when submitting bug reports. Example below: =========================== == SVN-branch-1.8-... == Currently Held Locks =========================== (closes issue ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a7732dfa30	Clear the DTMF sending digit tracking on off nominal paths ... In certain situations, when the RTP engine goes to send a DTMF end digit it may be in a situation where the remote address is no longer available, or the digit that was supposed to be sent is invalid. In such cases, we need to clear the RTP counters appropriately. Otherwise, when the RTP source is set again, we'll continue to think that we're in the middle of sending a DTMF digit, which can confuse the remote party (signficantly). (closes issue ASTERISK-21522) Reported by: Corey Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e6c986dd33	Prevent crash in 'sip show peers' when the number of peers on a system is large ... When you have lots of SIP peers (according to the issue reporter, around 3500), the 'sip show peers' CLI command or AMI action can crash due to a poorly placed string duplication that occurs on the stack. This patch refactors the command to not allocate the string on the stack, and handles the formatting of a single peer in a separate function call. (closes issue ASTERISK-21466) Reported by: Guillaume Knispel patches: fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8d1f50d957	Fix CDR not being created during an externally initiated blind transfer ... Way back when in the dark days of Asterisk 1.8.9, blind transferring a call in a context that included the 'h' extension would inadvertently execute the hangup code logic on the transferred channel. This was a "bad thing". The fix was to properly check for the softhangup flags on the channel and only execute the 'h' extension logic (and, in later versions, hangup handler logic) if the channel was well and truly dead (Jim). Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected that the channel was leaving the bridge (but not to die) caused some crucial snippet of CDR code, lying in ambush in the middle of the bridging code, to not get executed. This had the effect of blowing away one of the CDRs that is typically created during a blind transfer. While we live and die by the adage "don't touch CDRs in release branches", this was our bad. The attached patch restores the CDR behavior, and still manages to not run the 'h' extension during a blind transfer (at least not when it's supposed to). Thanks to Steve Davies for diagnosing this and providing a fix. Review: https://reviewboard.asterisk.org/r/2476 (closes issue ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	cf90e44d99	Use the proper lower bound when doing saturation arithmetic. ... 16 bit signed integers have a range of [-32768, 32768). The existing code was using the interval (-32768, 32768) instead. This patch fixes that. Review: https://reviewboard.asterisk.org/r/2479/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	eeaac15fe9	Modifying sounds/Makefile to pull down 1.4.24 core sounds ... 1.4.24 core sounds includes a full set of Italian prompts for core sounds and a fix for the missing voicemail prompts in the Russian language. (closes issue ASTERISK-19431) (closes issue ASTERISK-19721) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Olle Johansson	fa0b6a2fb5	Play periodic prompst for first call in a call queue ... Review: https://reviewboard.asterisk.org/r/2263/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8e5ffde816	Clean up memory leak in config file on off nominal paths when glob is allowed ... If a system allows for its usage, Asterisk will use glob to help parse Asterisk .conf files. The config file loading routine was leaking the memory allocated by the glob() routine when the config file was in an unmodified or invalid state. This patch properly calls globfree in those off nominal paths. (closes issue ASTERISK-21412) Reported by: Corey Farrell patches: config_glob_leak.patch uploaded by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d3b56c6ad4	Clean up resources in features on exit ... This patch cleans up two things features: * It properly unregisters the CLI commands that features registered * It cancels and performs a pthread_join on the created parking thread. This not only properly joins a non-detached thread, but also prevents disposing of the parking lots prior to the parking thread completely exiting. (closes issue ASTERISK-21407) Reported by: Corey Farrell patches: features_shutdown-r2.patch uploaded by Corey Farrell (License 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	edc8f9439a	Change Case On Forcerport For Consistency ... * Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386483 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b5c39ac861	Fix crash when AMI redirect action redirects two channels out of a bridge. ... The two party bridging loops were changing the bridge peer pointers without the channel locks held. Thus when ast_channel_massquerade() tested and used the pointer there is a small window of opportunity for the pointers to become NULL even though the masquerade code has the channels locked. (closes issue ASTERISK-21356) Reported by: William luke Patches: jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett Tested by: William luke git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6f8468d863	Prevent res_timing_pthread from blocking callers ... There were several reports of deadlock when using res_timing_pthread. Backtraces indicated that one thread was blocked waiting for the write to the pipe to complete and this thread held the container lock for the timers. Therefore any thread that wanted to create a new timer or read an existing timer would block waiting for either the timer lock or the container lock and deadlock ensued. This patch changes the way the pipe is used to eliminate this source of deadlocks: 1) The pipe is placed in non-blocking mode so that it would never block even if the following changes someone fail... 2) Instead of writing bytes into the pipe for each "tick" that's fired the pipe now has two states--signaled and unsignaled. If signaled, the pipe is hot and any pollers of the read side filedescriptor will be woken up. If unsigned the pipe is idle. This eliminates even the chance of filling up the pipe and reduces the potential overhead of calling unnecessary writes. 3) Since we're tracking the signaled / unsignaled state, we can eliminate the exta poll system call for every firing because we know that there is data to be read. (closes issue ASTERISK-21389) Reported by: Matt Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches: 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417) (closes issue ASTERISK-19754) Reported by: Nikola Ciprich (closes issue ASTERISK-20577) Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported by: Henry Fernandes (closes issue ASTERISK-17467) Reported by: isrl (closes issue ASTERISK-17458) Reported by: isrl Review: https://reviewboard.asterisk.org/r/2441/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	46af845e4a	cli.c: Properly initialize debug_modules and verbose_modules. ... This avoids some lock errors on the core set {debug,verbose} commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	e4854dd1b2	Distributed Device State broken at sites using res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is inplace ... res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in, then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	c41e19fc90	Don't unnecessarily rebuild things on every run of 'make'. ... Review: https://reviewboard.asterisk.org/r/2449/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a6d0dfc3d1	Fix the svn:keywords property on several files. ... Normally I think keyword expansion is silly, but the one time it would have been good, it didn't work because the property had quotes in it. This patch fixes obviously busted svn:keywords properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5a756f15f8	Calculate the timestamp for outbound RTP if we don't have timing information ... This patch calculates the timestamp for outbound RTP when we don't have timing information. This uses the same approach in res_rtp_asterisk. Thanks to both Pietro and Tzafrir for providing patches. (closes issue ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035) rtp-timestamp.patch uploaded by pbertera (License 5943) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a73ffd9d25	Don't attempt to create a voice frame on a read error ... Prior to this patch, a read error in snd_pcm_readi would still be treated as a nominal result when constructing a voice frame from the expected data. Since the value returned is negative, as opposed to the number of samples read, this could result in a crash. With this patch, we now return a null frame when a read error is detected. Note that the patch on ASTERISK-21329 was modified slightly for this commit, in that we bail immediately on detecting the read error, rather than bypassing the construction of the voice frame. (closes issue ASTERISK-21329) Reported by: Keiichiro Kawasaki patches: chan_alsa.diff uploaded by kawasaki (License 6489) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	7ff1e29675	Fix Manager Segfault When app_queue Is Unloaded ... When app_queue is unloaded, some manager commands are not being unregistered which result in a segfault. This patch corrects this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann, Corey Farrell Tested by: Corey Farrell Patches: asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L. Young (license 5026) asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2444/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	1f2145b7b4	Fix app_voicemail Segfault And A Few Memory Leaks ... The original report was that app_voicemail would crash. This was caused by ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being performed for that return status. After adding the initial patch to fix this issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered. During review, Walter Doekes (wdoekes) suggested adding a helper function in order to determine if we had a valid configuration or not. This patch does the following: * Creates a helper function to check if the configuration is valid * Adds calls to the new helper function where appropiate * Fixes memory leaks where the code returned without running ast_config_destroy() on the configuration that was loaded (closes issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco Kroon, Michael L. Young Patches: asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2443/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	1f89548d65	IAX2 defer_full_frames fail to get sent ... Ensure iax2_process_thread is signalled when a deferred frame is queued to it. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	e28a50dbb5	IAX2, prevent network thread starting before all helper threads are ready ... On startup, it's possible for a frame to arrive before the processing threads were ready. In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive before we are at ast_cond_wait, the signal will be ignored. The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued. Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled after each thread is started. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2427/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6166563bb3	Use LDAP memory management functions instead of Asterisk's ... When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk: invalid pointer errors) can occur as the memory is being allocated with Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the LDAP library's wrappers. This patch uses the LDAP library's wrappers where appropriate, so that compiling with MALLOC_DEBUG doesn't cause more problems than it solves. Note that the patch listed below was modified slightly for this commit to account for some additional memory allocation/deallocations. (closes issue ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham patches: issue18789-1.8-r316873.patch uploaded by seanbright (License 5060) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f39c2a40f9	Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE ... When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	ef5a22c5b0	Modified the list of keys for the driver backends for sake of sample clarity ... Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	68ebcb8f9d	Fix For Not Overriding The Default Settings In chan_sip ... The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. (closes issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2386/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	c5d2978944	Backport Appropiate NAT Setting Cleanup ... In ASTERISK-20904, the focus was around the changes to NAT that took place in Asterisk 11. Since the report stated that 1.8 was fine, we didn't take a look at 1.8 at the time. While working on ASTERISK-21225, I could see that 1.8 would benefit from having some of those changes applied to it. This patch does the following: * The important part of this patch is that it sets the peer's flags earlier in build_peer so that the code properly uses the peer's flags based on the peer's configuration. * constify req parameter in check_via() * update realtime schemas under the contrib directory to handle properly the NAT settings available in 1.8 as well as to handle the changes made in 11 to make upgrading easier when installing newer versions of Asterisk (closes issue ASTERISK-21243) Reported by: Michael L. Young Patches: asterisk-20904-changes_for_1.8.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2422/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7bdeb23dd2	chan_dahdi: Add inband_on_proceeding compatibility option. ... The new inband_on_proceeding option causes Asterisk to assume inband audio may be present when a PROCEEDING message is received. Q.931 Section 5.1.2 says the network cannot assume that the CPE side has attached to the B channel at this time without explicitly sending the progress indicator ie informing the CPE side to attach to the B channel for audio. However, some non-compliant ISDN switches send a PROCEEDING without the progress indicator ie indicating inband audio is available and assume that the CPE device has connected the media path for listening to ringback and other messages. ASTERISK-17834 which causes this issue was dealing with a non-compliant network switch. (closes issue ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago