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2012-07-09Document that multiple endpoints using the same connection is not supported.Joshua Colp
(closes issue ASTERISK-20104) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add Digium phones context to sip_notify sample config.Jason Parker
This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Fix an issue where media would not flow for situations where the legacy STUN ↵Joshua Colp
code is in use. The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20102) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add additional namespaces for Google Talk which are used for the gmail client.Joshua Colp
(closes issue ASTERISK-20101) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used ↵Joshua Colp
to use res_jabber. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09chan_sip: Fix small behavioral change accidentally introduced in r369750Jonathan Rose
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369793 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-07Add a new unified Jingle, Google Jingle, and Google Talk channel driver ↵Joshua Colp
written from scratch called chan_motif. This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either. These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold, unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications. The original Google Talk protocol is also supported for Google Voice interoperability. You may ask yourself though where the name motif comes from... and I would say to you... music! motif: a perceivable or salient recurring fragment or succession of notes Sorta like a jingle! Review: https://reviewboard.asterisk.org/r/1917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Remove unnecessary generation of informational cause framesKinsey Moore
It is not necessary to generate information cause code frames on every protocol event that occurs. This removes all the instances where the frame was not conveying a cause code and was instead just conveying a protocol-specific message. This also corrects the generation of the message associated with disconnects for MFC/R2 to use the MFC/R2 specific text for the disconnect cause. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06chan_sip: Add case for FLASH control frames so that we don't display a warning.Jonathan Rose
chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369751 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Remove a superfluous and dangerous freeing of an SSL_CTX.Mark Michelson
The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369732 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Fix bridging thread leak.Mark Michelson
The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369709 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06Import revision 4196 from pjproject trunk. Fix a crash issue when starting ↵Joshua Colp
ICE connectivity checks and immediately destroying the ICE session. This was exposed by the SIP CCSS test. Full fix for this issue will be worked on as a medium to long term roadmap item. pjroject issue viewable at https://trac.pjsip.org/repos/ticket/1548 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Add 'stun show status' commandMatthew Jordan
This patch adds a new CLI command, 'stun show status'. This command will show a table describing all known STUN servers and statuses. (closes issue ASTERISK-18046) Reported by: Jeremy Kister Tested by: Jeremy Kister patches: (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy Kister) Review: https://reviewboard.asterisk.org/r/2001 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Make res/pjproject ignore more files.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05AST-2012-011: Resolve heap corruption issue with voicemailKinsey Moore
The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369653 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Make res/pjproject ignore some generated files.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Tweak some comments and whitespace in utils.hRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05app_mixmonitor: Fix a reference leak in manager_mixmonitor functionJonathan Rose
Manager_mixmonitor included an early return on failed executions of mixmonitor that would result in a leaked channel reference. (closes issue ASTERISK-19943) Reported by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded by Mark Murawski (license 5791) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Do not send a BYE when a provisional response arrives during a re-INVITEMatthew Jordan
Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369627 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Fix dev mode ooh323 warningsAlexandr Anikin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-04Added direct media support to ooh323 channel driverAlexandr Anikin
options are documented in config sample sample config rename to proper name - ooh323.conf To change media address ooh323 send empty TCS if there was completed TCS exchange or send facility forwardedelements with new fast start proposal if not. Then close transmit logical channels and renew TCS exchange. If new fast start proposal is received then ooh323 stack call back channel driver routine to change rtp address in the rtp instance. If empty TCS is received then close transmit logical channels and renew TCS exchange Review: https://reviewboard.asterisk.org/r/1607/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-04fix small mistake in the previousAlexandr Anikin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-04Fix modern gcc warningAlexandr Anikin
Review: https://reviewboard.asterisk.org/r/1767 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-03More improvements to re-INVITEs timing out after a provisional responseTerry Wilson
There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369580 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-03Better handle re-INVITEs with provisional but no final repsonsesTerry Wilson
A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-02Add a cleaned up drop-in replacement for res_jabber called res_xmpp. This ↵Joshua Colp
provides the same externally facing functionality but is implemented differently internally. This is currently not built by default but this will be changed once chan_jingle2 (insert actual name in your head when reading this after it has been merged) is in the tree. Review: https://reviewboard.asterisk.org/r/1983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-02Ensure the timer heap is protected by a lock.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Enable IPv6 support in pjproject.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Don't try to send connectivity checks on RTCP if RTCP is no longer present ↵Joshua Colp
and don't do multiple ICE connectivity checks at once. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Fix apparent copy and paste error where incorrect "glue" is used.Mark Michelson
........ Merged revisions 369511 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Hangup handlers - Dialplan subroutines that run when the channel hangs up.Richard Mudgett
Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29With some configurations a transport is not actually specified so assume UDP ↵Joshua Colp
in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369491 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Remove obsolete struct ast_channel note.Richard Mudgett
The opaquing the ast_channel struct no longer requires .cleancount to be changed when the struct is changed. * Bump .cleancount value one last time because of struct ast_channel for old times sake. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Make the address family filter specific to the transport.Joshua Colp
(closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-28Add the ability to set flags via the config options apiTerry Wilson
Allows the setting of flags via the config options api. For example, code like this: #define OPT1 1 << 0 #define OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int flags; }; and a config like this: [blah] opt1=yes opt2=no opt3=yes Review: https://reviewboard.asterisk.org/r/2004/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-27AST-2012-010: Clean up after a reinvite that never gets a final responseTerry Wilson
The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26Unique Call ID logging Phases III and IVJonathan Rose
Adds call ID logging changes to specific channel drivers that weren't handled handled in phase II of Call ID Logging. Also covers logging for threads for threads created by systems that may be involved with many different calls. Extra special thanks to Richard for rigorous review of chan_dahdi and its various signalling modules. review: https://reviewboard.asterisk.org/r/1927/ review: https://reviewboard.asterisk.org/r/1950/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26Fix crash in unloading of res_adsi moduleMatthew Jordan
When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. ........ Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369391 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Update "manager show event" to support tab completionMatthew Jordan
Thank you rmudgett for pointing out that I was missing this in the initial check-in for AMI event documentation (r369346) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Fix incorrect duration reporting in CDRs created in batch modeMatthew Jordan
Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ ........ Merged revisions 369351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Re-fix how local tag is generated when sending a 481 to an INVITE.Mark Michelson
Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Add AMI event documentationMatthew Jordan
This patch adds the core changes necessary to support AMI event documentation in the source files of Asterisk, and adds documentation to those AMI events defined in the core application modules. Event documentation is built from the source by two new python scripts, located in build_tools: get_documentation.py and post_process_documentation.py. The get_documentation.py script mirrors the actions of the existing AWK get_documentation scripts, except that it will scan the entirety of a source file for Asterisk documentation. Upon encountering it, if the documentation happens to be an AMI event, it will attempt to extract information about the event directly from the manager event macro calls that raise the event. The post_process_documentation.py script combines manager event instances that are the same event but documented in multiple source files. It generates the final core-[lang].xml file. As this process can take longer to complete than a typical 'make all', it is only performed if a new make target, 'full', is chosen. Review: https://reviewboard.asterisk.org/r/1967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Fix Bridge application occasionally returning to the wrong location.Richard Mudgett
* Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque ........ Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369328 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Multiple revisions 369323-369324Mark Michelson
........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module. The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 ........ r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines Forgot to svn add this file in my last commit. ........ Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369325 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Be more consistent with the return code for requests received from invalid ↵Mark Michelson
domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369303 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-23Fix F and F(x) action logic in Bridge application.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-23Fix Bridge application and AMI Bridge action error handling.Richard Mudgett
* Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. ........ Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369283 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Explicitly check caller hangup in app Queue rather than a polluted res2 value.Richard Mudgett
........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369263 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Fix F and F(x) action logic in Queue application.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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