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2009-11-30Merged revisions 231614 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30Reverted 231616Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30Merged revisions 231614 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30app_queue crashes randomly, often during call-transfersDavid Vossel
This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-271. Modified exported variable names.TransNexus OSP Development
2. Added destination port support. 3. Added new protocols. 4. Added QoS. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-24Found a few places where queue refcounts were counted incorrectly. Also add ↵Tilghman Lesher
debug statements. (closes issue #15982, closes issue #15984) Reported by: atis Patches: 20091111__issue15982.diff.txt uploaded by tilghman (license 14) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-23Add an option to app_followme to disable the "please hold" announcement.Matthew Nicholson
(closes issue #14155) Reported by: junky Patches: M14555-trunk.diff uploaded by junky (license 177) (modified) Tested by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-20Fix/Implement error events for non-existing filesDavid Ruggles
also include a better cmd define for S command Review: https://reviewboard.asterisk.org/r/430/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-19Merged revisions 230508 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-19Basic cleanup of ExternalIVR: cleaned up argument parsing; implemented good ↵David Ruggles
coding practices where applicable; replaced most notice level logging with verbose logging; replaced warning messages that terminated with error messages; fixed memory leak identified by russellb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-16Fix another buglet in T.38 session teardown at the end of FAX sessions.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-16Ensure that only one end of a T.38 session initiates teardown at completion.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-161. Added SIP Diversion support.TransNexus OSP Development
2. Fixed compile warning for UUID. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Display a list of channel variables in each channel-oriented event.Tilghman Lesher
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Store the cause code that is returned when trying to create a channel in ↵Joshua Colp
ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of overwriting the device state in AVAILSTATUS. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Added full number portability parameter support.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-11Flags not initialized in app_softhangup.c, causing undefined behaviorDavid Brooks
Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-10When GOSUB is invoked within an AGI, it may not exit correctly.Tilghman Lesher
(closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-09Add the 'relative-periodic-announce' option to app_queue to allow for ↵Matthew Nicholson
calculating the time of announcments from the end of the previous announcment rather than from the beginning. (closes issue #15260) Reported by: tonils git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05Yet another error message in the dialplan (thanks, rmudgett/russellb)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05MEETME_INFO should not return a literal error message to the dialplan.Tilghman Lesher
(closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05Fix the fix for chanspy option o Jeff Peeler
In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05Don't crash if no arguments are passed.Tilghman Lesher
(closes issue #16119) Reported by: thedavidfactor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Added the 'a' option to app dial and modified app_dial to set the answertime ↵Matthew Nicholson
when the called channel answers. This change causes answertime to be correct even if the called channel hangs up during an announcement triggered by the A() option. (closes issue #15936) Reported by: falves11 Patches: dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96) dial-caller-answer1.diff uploaded by mnicholson (license 96) Tested by: falves11, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Merged revisions 227827 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Don't crash when state_interface is NULL.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03This patch adds a sequence field to CDRs that can be combined with the ↵Matthew Nicholson
linkedid or uniqueid field to uniquely identify a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03Add support for using a hint when configuring a state interface using the ↵Joshua Colp
format hint:<extension>@<context>. (closes issue #15168) Reported by: p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03Change warning message to debug message.Leif Madsen
app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02Merged revisions 226889 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22SIP TCP/TLS: move client connection setup/write into tcp helper thread, ↵David Vossel
various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22Permit storage of voicemail secrets in a separate file, located within the ↵Tilghman Lesher
spool directory. (closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22Merged revisions 225105 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ↵Kevin P. Fleming
ast_channel_iterator to use it. This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Apparently, I don't need to specify the ".so" suffix to get a matchTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Turn on DENOISE filter for all conference participants.Tilghman Lesher
(Fixes SWP-238) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Merged revisions 224565 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Allow ODBC storage to be queried with multiple mailboxes, and remove ↵Tilghman Lesher
multiple goto's. This corrects an issue reported on the -users list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15Readd removed ability to allow listening to one side of the call in app_chanspyJeff Peeler
(Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Revert inadvertant code commit to app_originateTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix handling of notification calls w/ the dialing apiTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Merged revisions 223804 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Remove automatic switching from T.38 to voice mode in chan_sip.Kevin P. Fleming
chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09Initiate T.38 switchover when acting as called party, regardless of FAX ↵Kevin P. Fleming
direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09Fix potential memory leaks.Mark Michelson
ABE-1998 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-06Recorded merge of revisions 222152 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Prevents from division by zeroMatthias Nick
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Modify VoiceMailMain()'s a() argument to allow mailboxes to be specified by ↵Sean Bright
name. (closes issue #14740) Reported by: pj Patches: issue14740_09022009.diff uploaded by seanbright (license 71) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Clarify documentation for VoiceMailMain()'s a() option.Sean Bright
We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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