asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	9e28dc510d	Fix memory leak when CEL is successfully written to PostgreSQL database PQClear is not called when the result object of a call to PQExec has a status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was handled properly, so this memory leak only occurred when CEL records were successfully written. This patch properly clears the result in the nominal code path. (closes issue ASTERISK-19991) Reported by: Etienne Lessard Tested by: Etienne Lessard patches: mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	5d843ed6fc	Prevent crash on shutdown due to refcount error on queues container. When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	90d111988e	Help prevent ringing queue members from being rung when ringinuse set to no. Queue member status would not always get updated properly when the member was called, thus resulting in the member getting multiple calls. With this change, we update the member's status at the time of calling, and we also check to make sure the member is still available to take the call before placing an outbound call. (closes issue ASTERISK-16115) reported by nik600 Patches: app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7c3ee0b760	AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers When an IAX2 call is made using the credentials of a peer defined in a dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are not applied to the call attempt. This allows for a remote attacker who is aware of a peer's credentials to bypass the ACL rules set for that peer. This patch ensures that the ACLs are applied for all peers, regardless of their storage mechanism. (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by: mjordan, Alan Frisch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	0196006388	AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR The AMI Originate action can allow a remote user to specify information that can be used to execute shell commands on the system hosting Asterisk. This can result in an unwanted escalation of permissions, as the Originate action, which requires the "originate" class authorization, can be used to perform actions that would typically require the "system" class authorization. Previous attempts to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought to do so by inspecting the names of applications and functions passed in with the Originate action and, if those applications/functions matched a predefined set of values, rejecting the command if the user lacked the "system" class authorization. As noted by IBM X-Force Research, the "ExternalIVR" application is not listed in the predefined set of values. The solution for this particular vulnerability is to include the "ExternalIVR" application in the set of defined applications/functions that require "system" class authorization. Unfortunately, the approach of inspecting fields in the Originate action against known applications/functions has a significant flaw. The predefined set of values can be bypassed by creative use of the Originate action or by certain dialplan configurations, which is beyond the ability of Asterisk to analyze at run-time. Attempting to work around these scenarios would result in severely restricting the applications or functions and prevent their usage for legitimate means. As such, any additional security vulnerabilities, where an application/function that would normally require the "system" class authorization can be executed by users with the "originate" class authorization, will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has been updated to reflect that the AMI Originate action can result in commands requiring the "system" class authorization to be executed. Proper system configuration can limit the impact of such scenarios. (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM X-Force Research git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	5eff857154	Restore CODING-GUIDELINES to doc folder In r294740, the CODING-GUIDELINES was removed from the doc folder in favor of the content on the Asterisk wiki. Some folks still look in the doc folder initially for coding guideline suggestions; as such, this patch adds a CODING-GUIDELINES file back into the doc folder. The content of the file merely points to the correct page on the Asterisk wiki where the coding guidelines currently live. (closes issue ASTERISK-20279) Reported by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	466e22fafc	app_meetme: Adding test events for following activity in MeetMe. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec030afdab	Initialize file descriptors for dummy channels to -1. Dummy channels usually aren't read from, but functions like SHELL and CURL use autoservice on the channel. (closes issue ASTERISK-20283) Reported by: Gareth Palmer Patches: svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2ed8fb7637	Fix hangup cause passthrough regression. The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	cf01ad5379	chan_sip: Send 408 on retransmit timeout instead of 603 (closes issue ASTERISK-20124) Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	97b5071653	Fix misleading documentation in agents.conf.sample regarding ackcall usage. The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	25e6cee370	Fix incorrect documentation of the MailboxStatus manager command. The "Waiting" field was misdocumented as reporting the number of messages waiting. In reality, it simply indicated the presence or absence of waiting messages. (closes issue AST-975) reported by John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	80834d503e	Fix incorrectly documented option in queues.conf sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David M. Lee	c977fe4197	Fixes ast_rwlock_timed[rd\|wr]lock for BSD and variants. The original implementations simply wrap pthread functions, which take absolute time as an argument. The spinlock version for systems without those functions treated the argument as a delta. This patch fixes the spinlock version to be consistent with the pthread version. (closes issue ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch uploaded by Egor Gorlin (license 6416) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	764fd6063d	Implement workaround for BETTER_BACKTRACES crash When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (closes issue ASTERISK-20090) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	d55de7831d	mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	3b476dfd1e	Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ccd68a5790	Ignore recovered zero-length secondary UDPTL packets In some cases, recovering lost packets using the secondary packet recovery mechanism with UDPTL/T.38 can result in the recovery of zero-length packets. These must be ignored or the frame generated from them can cause segfaults and allocation failures. (closes issue ASTERISK-19762) (closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	1ebccde21a	Fix memory leak in XML documentation When formatting documentation fields, the XML documentation parser calls xmldoc_get_formatted. This function allocates a string buffer at the beginning of its routine. Unfortunately, on certain code paths, it also calls xmldoc_string_cleanup, which assumes that it will create the string buffer. The previously allocated string buffer is then leaked by the xmldoc_string_cleanup routine. Now: we don't do that. (closes issue AST-932) Reported by: Alexander Homig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	309d55382d	Add instrumentation to subsystem reloads When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f817b8cdcd	Add module reload instrumentation for TEST_FRAMEWORK This adds AMI events for module reloads when Asterisk is built with TEST_FRAMEWORK enabled and corrects generation of the module load AMI event. (issue PQ-1126) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3fe0a3cfe5	Handle integer over/under-flow in ast_parse_args The strtol family of functions will return _MIN/_MAX on overflow. To detect when an overflow has happened, errno must be set to 0 before calling the function, then checked afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	c9bc743ff1	chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag was used instead, which will frequently flip during reinvites. (closes issue AST-897) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1393623e9b	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b0d337d2cb	Fix bug where final queue member would not be removed from memory. If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	01780bfda2	Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	af74988006	Add test instrumentation This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	4e0e1535c9	Fix problem where incorrect pointer was checked for nullity. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	53f102ed7c	Fix a couple of documentation problems in app_queue.c * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	eefc33dd34	remove ALREADYGONE flag on ooh323 call data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	c6d75bc2f5	Send re-register packets by GRQ (gatekeeper request) interval (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	af738476ca	Use better libss7 detection test and move libpri compile test. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	74d2044626	Fix to resend GRQ/RRQ if RRJ (registration reject) is received (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	ec6801aa0d	change opening h323 logfile with append mode instead of overwrite git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	483223b4d3	Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	38bbed178e	Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	13c82a2b30	Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b887a43667	Fix the analog dial *0 flash-hook of bridged peer feature. The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f051750307	Add missing AST_CAUSE_* -> text translations git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	7f3839ae03	Improve debug message for temporary outbound proxies. Thanks to Paul Belanger for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	9d402e02a5	Seriously? Another compilation error fixed. Somebody beat me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	dca8a1b14e	Remove unused variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	1119a574ed	Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	6a198f0045	Revert alloca changes for utils These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f1cd5b6e93	Schedule pokes of registered SIP peers within a given timespan after SIP reload With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	377caa7fb1	Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	80efa31733	Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	395b4b4898	Release B channel allocation on error path in chan_misdn. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e0c0f9934d	res_agi: Add message indicating need for \n character in verbose message The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	cb4bdd6531	Rewrite a comment that didn't adequately explain the code it was documenting. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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