asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	3c9667ae12	Merge queue-reset branch to Asterisk From a user point-of-view, this adds new CLI commands and Manager Actions to better facilitate the reloading of queues and the resetting of their statistics. The new CLI commands are the "queue reload" and "queue reset stats" commands. The new manager actions are the QueueReload and QueueReset commands. Review: http://reviewboard.digium.com/r/115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b5bde28d9f	Add manager events for chanspy starting or stopping (closes issue #14469) Reported by: caio1982 Patches: chanspy_events2.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	10281bf183	Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 \| mmichelson \| 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) \| 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0cfdaa2dc0	Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0d5da5f436	Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	34161542e9	Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	47ebea6a8d	Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c9670b24a4	Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d4df1c6a78	Fix0ring build git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	51c80f5b40	Remove the usage of the KeepAlive app, as it no longer exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	3084e9ce93	This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. (closes issue #14435) Reported by: D_McNaul git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	43462557c0	More intptr_t work. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	dbe56b6647	Merged revisions 174369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 \| murf \| 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) \| 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	829b14fc3e	Fixes issue with hangups not being sent and external process never terminating. The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b2484d7db9	Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8f9c8f8b40	Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	4ac807845d	Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 \| jpeeler \| 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) \| 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	2e1d9f9a21	Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 \| mmichelson \| 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) \| 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5d3fc1ee3a	Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 \| mmichelson \| 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) \| 3 lines Add some missing cleanup to app_mixmonitor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5f83636af4	Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 \| mmichelson \| 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) \| 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	172777bd02	Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e04e940a71	Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 \| mmichelson \| 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) \| 3 lines Revert my previous change because it was stupid ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	668f9f4e31	Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 \| mmichelson \| 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) \| 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e179e613f7	1. Make OS X compile cleanly with app_stack. 2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	53d9b77898	This reverts the changes I made for 11583; will	17 years ago
Steve Murphy	c61e8a7865	This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	317e4e7313	Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	8d782f96b8	Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 \| twilson \| 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) \| 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	80f91f6ab7	Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 \| tilghman \| 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) \| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	7ecda45482	Fix "cancel answered elsewhere" through app_queue with members in chan_local. Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	268ac221a2	Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 \| murf \| 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) \| 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	097822966b	Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on. This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	04e56bde03	Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	ec5aa59105	Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 \| seanbright \| 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) \| 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	665bba38f1	Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 \| file \| 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) \| 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fcf4d42cde	Reset the ast_str used for escape substitution. We need to do this since it is a thread local variable that may contain the value of a previous substitution. (closes issue #14312) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f6f5197e63	Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 \| file \| 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) \| 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	9ee6a741dc	Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4263503bd5	Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b52253d590	Use the default timeout for a queue instead of -1 (closes issue #14272) Reported by: timking git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	071301029c	Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 \| tilghman \| 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) \| 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3728c3aa92	Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 \| tilghman \| 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) \| 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9a7efae8fd	remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC this stops modules from being linked against both sets of libraries on systems that have both installed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	fcb69e6f9d	Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	526cc089a9	Add support for setting the Reason header when cancelling a call in the queue because someone else answered. Previously, only dial() was supported. EDV-102 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b9060d4435	Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 \| mmichelson \| 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) \| 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a474341105	Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 \| murf \| 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) \| 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	2015f9c980	Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 \| twilson \| 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) \| 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ec0f18405e	Clarify a message that app_queue prints and change to a debug-level message The "No one is answering..." verbose message contained 3 numbers that were not explained in any way to whoever was viewing the message. It is more helpful now since the message explains what the numbers mean. Also, the message has been downgraded to "DEBUG" level. (closes issue #14172) Reported by: caio1982 Patches: queue_answering_debug.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 \| russell \| 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) \| 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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3377 Commits (0a792331bff9bf6568d485030c7a214e20285d2c)