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${ noResults }
22709 Commits (3805e2ae4d654a24f7c98fc6f41dcec81d02670e)
Author | SHA1 | Message | Date |
---|---|---|---|
|
3805e2ae4d |
Fix failing SDP_offer_answer test
Asterisk now generates image stream declinations with the same transport case that it used to before the stream declination improvements. (udptl vs UDPTL) (closes issue SWP-4736) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369900 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
55871d3a67 |
Add additional description stanza names from the old Google Talk protocol which is used with Google Voice.
(closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
74ebe6d5ab |
Respect codec preference order when adding codecs to a media description.
This change allows an endpoint in motif.conf to be configured with a preference of G.722 and fallback of ulaw. With Google this allows communication with Google Talk clients to use G.722 while when using Google Voice ulaw will be used. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
6416a246ed |
Improve Goto and GotoIf related documentation
Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
b1bb826350 |
Fix initial loading problem with res_curl
When the OpenSSL duplicate initialization issues were resolved in r351447, res_curl could fail to load if it checked SSL_library_init after SSL initialization completed. This is due to the SSL_library_init stub returning a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of 1 to indicate success - in fact, SSL_library_init is documented to always return 1. Interestingly, the CURL libraries actually checked the return value - the fact that nothing else that depends on OpenSSL was having problems loading probably means they don't check the return value. (closes issue AST-924) Reported by: Guenther Kelleter patches: (AST-924.patch license #6372 uploaded by Guenther Kelleter) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
7296b670d4 |
Add required items for Google video support.
This adds legacy STUN support for RTCP sockets, adds RTCP candidates to the Google transport information, and adds required codec parameters. (closes issue ASTERISK-20106) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8f162be802 |
When receiving a STUN binding request send one out as the Google Talk client uses this as a method to determine if the remote party is still reachable or not.
Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis. (closes issue ASTERISK-20107) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
7baa8bf43d |
Add support for exposing the received contact URI and also for setting the request URI in messages.
(closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
b46e1b45e4 |
Force the clock rate of G.722 to be 16000 when using the Google transports as it is 8000 elsewhere.
(closes issue ASTERISK-20105) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
04504e80a3 |
Document that multiple endpoints using the same connection is not supported.
(closes issue ASTERISK-20104) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
b1cd995273 |
Add Digium phones context to sip_notify sample config.
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 |
13 years ago |
|
31beb35f47 |
Fix an issue where media would not flow for situations where the legacy STUN 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 |
13 years ago |
|
540f4b81f9 |
Add additional namespaces for Google Talk which are used for the gmail client.
(closes issue ASTERISK-20101) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
fa0bcb6c70 |
Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used to use res_jabber.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
60bc927579 |
chan_sip: Fix small behavioral change accidentally introduced in r369750
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 |
13 years ago |
|
a3fa37b8cf |
Add a new unified Jingle, Google Jingle, and Google Talk channel driver 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 |
13 years ago |
|
db59a3f123 |
Remove unnecessary generation of informational cause frames
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 |
13 years ago |
|
49aa47171b |
chan_sip: Add case for FLASH control frames so that we don't display a warning.
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 |
13 years ago |
|
8260fdfdd1 |
Remove a superfluous and dangerous freeing of an SSL_CTX.
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 |
13 years ago |
|
8e7ad68b1a |
Fix bridging thread leak.
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 |
13 years ago |
|
96a4b257bd |
Import revision 4196 from pjproject trunk. Fix a crash issue when starting 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 |
13 years ago |
|
3044aa3e38 |
Add 'stun show status' command
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 |
13 years ago |
|
6a1ec1c208 |
Make res/pjproject ignore more files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
163d3b05d4 |
AST-2012-011: Resolve heap corruption issue with voicemail
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 |
13 years ago |
|
f653a2026d |
Make res/pjproject ignore some generated files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1906601bf4 |
Tweak some comments and whitespace in utils.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
70e34d3354 |
app_mixmonitor: Fix a reference leak in manager_mixmonitor function
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 |
13 years ago |
|
4b3476d016 |
Do not send a BYE when a provisional response arrives during a re-INVITE
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 |
13 years ago |
|
f719be6054 |
Fix dev mode ooh323 warnings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
fa10f3f8a8 |
Added direct media support to ooh323 channel driver
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 |
13 years ago |
|
50765000e6 |
fix small mistake in the previous
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
324e47342e |
Fix modern gcc warning
Review: https://reviewboard.asterisk.org/r/1767 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
474b023ad4 |
More improvements to re-INVITEs timing out after a provisional response
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 |
13 years ago |
|
d97e6c1401 |
Better handle re-INVITEs with provisional but no final repsonses
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 |
13 years ago |
|
213bbc169a |
Add a cleaned up drop-in replacement for res_jabber called res_xmpp. This 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 |
13 years ago |
|
c48d346d55 |
Ensure the timer heap is protected by a lock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
09eb252721 |
Enable IPv6 support in pjproject.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
3f9cfe2d41 |
Don't try to send connectivity checks on RTCP if RTCP is no longer present 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 |
13 years ago |
|
37256ea45d |
Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
628425ba6f |
Fix apparent copy and paste error where incorrect "glue" is used.
........ 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 |
13 years ago |
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ac35b92b62 |
Hangup handlers - Dialplan subroutines that run when the channel hangs up.
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 |
13 years ago |
|
35c533156c |
With some configurations a transport is not actually specified so assume UDP 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 |
13 years ago |
|
6681e88bdd |
Remove obsolete struct ast_channel note.
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 |
13 years ago |
|
2e23dbb4b6 |
Make the address family filter specific to the transport.
(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 |
13 years ago |
|
1609fca6bb |
Add the ability to set flags via the config options api
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 |
13 years ago |
|
7d9e0158c3 |
AST-2012-010: Clean up after a reinvite that never gets a final response
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 |
13 years ago |
|
5eb94d7ebb |
Unique Call ID logging Phases III and IV
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 |
13 years ago |
|
ee11118695 |
Fix crash in unloading of res_adsi module
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 |
13 years ago |
|
5d31fb2dd2 |
Update "manager show event" to support tab completion
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 |
13 years ago |
|
bebdbf3381 |
Fix incorrect duration reporting in CDRs created in batch mode
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 |
13 years ago |