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2010-02-01Backup and restore original textfile, for prosthesis (gerund of prepend).tilghman1-6/+30
Also, fix menuselect such that changing voicemail build options correctly causes rebuild. (closes issue #16415) Reported by: tomo1657 Patches: prepention.patch uploaded by tomo1657 (license 484) (with modifications by me to backport to 1.4) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@244242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Revert 243570, I should have looked at this closer. Will reopen the issue, butjpeeler1-1/+1
am leaving the review closed as the change was pointless. (issue #16488) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243691 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Extend announcement URL used with Queue from 80 chars to PATH_MAX.jpeeler1-1/+1
(closes issue #16488) Reported by: syspert Patches: soundfilelen.pacth-2 uploaded by syspert (license 938) Review: https://reviewboard.asterisk.org/r/475/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-15Disallow leaving more than maxmsg voicemails.tilghman1-5/+72
This is a possibility because our previous method assumed that no messages are left in parallel, which is not a safe assumption. Due to the vmu structure duplication, it was necessary to track in-process messages via a separate structure. If at some point, we switch vmu to an ao2-reference-counted structure, which would eliminate the prior noted duplication of structures, then we could incorporate this new in-process structure directly into vmu. (closes issue #16271) Reported by: sohosys Patches: 20100108__issue16271.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: jsutton git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@240414 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13add silence gen to wait appsdvossel2-0/+17
asterisk.conf's 'transmit_silence' option existed before this patch, but was limited to only generating silence while recording and sending DTMF. Now enabling the transmit_silence option generates silence during wait times as well. To achieve this, ast_safe_sleep has been modified to generate silence anytime no other generators are present and transmit_silence is enabled. Wait apps not using ast_safe_sleep now generate silence when transmit_silence is enabled as well. (closes issue 0016524) Reported by: kobaz (closes issue 0016523) Reported by: kobaz Tested by: dvossel Review: https://reviewboard.asterisk.org/r/456/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@239718 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Resolve a crash due to an ast_frame not being fully initialized.russell1-2/+4
(closes issue #16531) Reported by: john8675309 (closes SWP-615) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238009 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Avoid a crash with large numbers of MeetMe conferences.seanbright1-30/+33
Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236509 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-11Fix talking detection status after conference user is muted.jpeeler1-20/+38
This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234379 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Update chan_spy documentation.lmadsen1-2/+2
Update the documentation in ChanSpy and ExtenSpy to reflect that only a single group can be specified to the g() option. (closes issue #16420) Reported by: diatonic git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-04document and rename strip_control() in app_voicemaildvossel1-3/+9
(closes issue #16291) Reported by: wdoekes git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233116 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Deprecate "cz" in favor of "cs".tilghman1-63/+75
Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232820 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02fixes app_queue ao2 errordvossel1-2/+2
(closes issue #16369) Reported by: vrban Patches: queue_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232444 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Fix a bug where if you hung up very quickly after calling AMD it would ↵file1-0/+1
overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Remove duplicate entries from voicemail format lists. This prevents ↵mnicholson1-2/+11
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: http://svn.digium.com/svn/asterisk/branches/1.4@231614 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30app_queue crashes randomly, often during call-transfersdvossel1-100/+114
In app_queue, it is possible for a call_queue to be destroyed while another object still holds a pointer to it. This patch converts call_queue objects to ao2 objects allowing them to be ref counted. This makes it safe for the queue_ent object in queue_exec() to reference it's parent call_queue even after it has left the queue. (closes issue #15686) Reported by: Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license 671) Tested by: dvossel, aragon Review: https://reviewboard.asterisk.org/r/427/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231437 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-25fixes solaris segfault on dial with verbosity >= 3dvossel1-2/+2
(closes issue #16193) Reported by: asgaroth Patches: bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231235 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-19fixes MixMonitor thread not exiting when StopMixMonitor is useddvossel1-28/+70
(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: http://svn.digium.com/svn/asterisk/branches/1.4@230508 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Document a limitation in the AVAILSTATUS variable from ChanIsAvail and providefile1-1/+6
a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229965 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04This patch modifies the Dial application to monitor the calling channel for ↵mnicholson1-13/+62
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: http://svn.digium.com/svn/asterisk/branches/1.4@227827 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Fix a bug where the recorded privacy introduction file would not get removed ↵file1-14/+31
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: http://svn.digium.com/svn/asterisk/branches/1.4@226889 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Fix documentation for ast_softhangup() and correct the misuse thereof.tilghman1-1/+1
(closes issue #16103) Reported by: majorbloodnok git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225105 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Suffix is not needed for a matchtilghman1-16/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225103 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Do not attempt early media bridging (ie: direct RTP setup) if options are ↵file1-6/+11
enabled that should prevent it. (closes issue #14763) Reported by: cupotka git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224565 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Ensure ringing continues for branched calls after progress is receivedjpeeler1-2/+7
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: http://svn.digium.com/svn/asterisk/branches/1.4@223804 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-11Remove a duplicate ao2_iterator_destroy().russell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223550 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Fix potential memory leak in app_dial.cmmichelson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223213 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Fix ao2_iterator API to hold references to containers being iterated.kpfleming1-2/+29
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: http://svn.digium.com/svn/asterisk/branches/1.4@222152 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-29Avoid a deadlock in chanspy, just in case the spyee is masqueraded and ↵mnicholson1-3/+4
chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@220907 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-24Implicitly sending a progress signal breaks some applications.tilghman2-9/+1
Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@220288 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-22When IMAP variables were changed during a reload, Voicemail did not use the ↵tilghman1-2/+22
new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@219816 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15If the user enters the same password as before, don't signal an error when ↵tilghman1-8/+7
the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218730 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Ensure FollowMe sets language in channels it creates.tilghman1-0/+4
Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218577 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-14Don't say "Please try again" if we don't give the user another chance to try ↵tilghman1-0/+6
again. (issue #15055, SWP-129) Reported by: jthurman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218331 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-14Ensure we don't pickup ourselves when doing pickup by exten.mnicholson1-1/+1
(closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218223 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Don't ring another channel, if there's not enough time for a queue member to ↵tilghman1-4/+21
answer. (Fixes AST-228) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@217989 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08When MOH is playing on the channel, announcements sent through the ↵tilghman1-12/+30
conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@217156 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-04Make apps send PROGRESS control frame for early media and fix too early ↵oej2-2/+11
media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@216430 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-01Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel namesdhubbard1-3/+5
In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@215270 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-20Make all the symbols for the C-client callbacks globaljpeeler1-0/+20
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@213283 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Fixes memory leak caused by incorrectly freeing mixmonitordvossel1-2/+2
(closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@213103 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12This patch adds additional checking when generating queue log TRANSFER events.mnicholson1-1/+3
The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211953 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman25-99/+111
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211528 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-07Resolve a deadlock involving app_chanspy and masquerades.russell1-6/+10
(ABE-1936) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211112 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-07QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername.tilghman1-2/+2
This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211038 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Reverting index() fix, applying a different methodology, based upon ↵tilghman1-4/+4
developer discussions. (related to issue #15639) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@210066 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-01Modify how Playtones() is used in Milliwatt() to resolve gain issue.russell1-7/+3
When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@209838 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Don't impose an arbitrary limit on member lines in queues.confmmichelson1-2/+5
I know what some of you are thinking: "UGH! Mark, why are you using ast_strdup and ast_free for the string when you can just use ast_strdupa and let the memory free itself?! Have the bats been chewing on your brain again?" Based on past experiences, I don't like using ast_strdupa inside a loop. It's a good way to potentially exhaust stack space. Also, since this only happens when reloading queues, I don't think that heap allocations and frees are going to be a huge problem. (closes issue #15559) Reported by: amorsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@208622 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Do not log an ERROR if autoservice_stop() returns -1.russell1-1/+0
This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@208592 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Prevent phantom calls to queue members.mmichelson1-2/+3
If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@205349 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29Place unlock of mutex in an else block so that it does not get unlocked twice.mmichelson1-1/+2
(closes issue #15400) Reported by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@204012 f38db490-d61c-443f-a65b-d21fe96a405b
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