asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	f68807f063	Remove most of the contents of the doc dir in favor of the wiki content. This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	01f31e0c50	Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 \| jpeeler \| 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) \| 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 \| jpeeler \| 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) \| 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	aafc5f4e99	Merged revisions 294639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294639 \| jpeeler \| 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) \| 53 lines Merged revisions 294384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 \| jpeeler \| 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) \| 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	c6973b1187	Fixing the Mac OS X build (bamboo warning) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1c692b86e5	Properly queue files with inotify(7). (closes issue #18089) Reported by: abelbeck Patches: 20101021__issue18089.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a4715fdf2d	Tweak a couple of CLI commands back to their original form. The "module" in this case is two parts, so there are two words before the verb of the CLI command. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2ee0c4ddff	Merged revisions 294500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294500 \| russell \| 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) \| 7 lines Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1334febddc	Allow ast_do_masquerade() failure to be reported again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2d6a9d607f	Merged revisions 294429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 \| tilghman \| 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) \| 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3f9644b7db	Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b12f27f231	Merged revisions 294312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294312 \| jpeeler \| 2010-11-08 16:30:49 -0600 (Mon, 08 Nov 2010) \| 1 line add missing unlock not present in 294277 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	375b5b6b09	Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 \| jpeeler \| 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) \| 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	529b8fc988	Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 \| mnicholson \| 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) \| 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	3bb6f58420	Set a default waittime, and make sure to convert it to milliseconds git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4e39e38ac5	valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ******** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ******** JIRA SWP-2423 JIRA ABE-2614 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	e1a1360451	Fixed deadlock avoidance issues while locking channel when adding the Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	225da545c5	Corret spelling and example git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	10317dacb6	Tell people to use the correct common name for clients as well git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Shaun Ruffell	c45ac8cc9c	Merged revisions 293969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293969 \| sruffell \| 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) \| 25 lines Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 \| sruffell \| 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) \| 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c34f40e710	Fixes ringback tone on sip semi-attended transfer. ABE-2168 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5cf1c93fd5	Do not output port in IPaddress for AMI sippeers. (closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	252cffd38d	Merged revisions 293806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 \| rmudgett \| 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) \| 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 \| rmudgett \| 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) \| 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6dfc9dddd8	Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9a482b0724	Merged revisions 293723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 \| jpeeler \| 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) \| 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 \| jpeeler \| 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) \| 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	643d500d66	Merged revisions 293647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 \| rmudgett \| 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) \| 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 \| rmudgett \| 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) \| 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	2c98e867f4	If manager and tls are disabled, do not display TCP/TLS Bindaddress. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ea00bff969	Analog 3-way call would not connect all parties if one was using sig_pri. Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8b53a54dc3	Use ast_sockaddr_from_sin function not memcpy This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	de440fa10d	Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	651d43c698	Merged revisions 293417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 \| rmudgett \| 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 \| rmudgett \| 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some more code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	55a1b13da8	Merged revisions 293340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 \| rmudgett \| 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 \| rmudgett \| 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	da39c33809	Modify sip_setoption to not complain about unknown options. This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e90dbe2963	Merged revisions 293195-293196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293195 \| tilghman \| 2010-10-28 14:52:52 -0500 (Thu, 28 Oct 2010) \| 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 \| tilghman \| 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) \| 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ r293196 \| tilghman \| 2010-10-28 14:54:34 -0500 (Thu, 28 Oct 2010) \| 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 \| tilghman \| 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) \| 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	971db9fc4e	Merged revisions 293158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 \| jpeeler \| 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) \| 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6cbda6ed92	Merged revisions 293118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 \| jpeeler \| 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) \| 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 \| jpeeler \| 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) \| 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	70415ccdfd	No need to define the struct if there are no users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d411ade931	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d1aceac167	Several more defines that need to be altered for compiling against an older version of libpri git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cdc420235e	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3020818583	Merged revisions 292867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 \| dvossel \| 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) \| 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 \| dvossel \| 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) \| 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	739dcea39b	Don't create directories without at least o+x Also, making files that you are going to modify read-only is dumb. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	af6a9e6199	Make files readable only by the owner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	9baf979137	Merged revisions 292786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 \| lmadsen \| 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) \| 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fff80bbe00	Prevent multiple runs of event_sub_test from producing false failure results. The array of test subscriptions was declared "static," meaning that the data.count field would retain its value between runs of the test. After the first test run, this would result in false reports of test failures. I chose to just remove the "static" keyword from the structure since it's not a huge deal to construct this structure during each run of the test. Another alternative would have been to zero out the data.count fields of each test subscription instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	177d65f59c	Add TLS cert helper script This script is useful for quickly generating self-signed CA, server, and client certificates for use with Asterisk. It is still recommended to obtain certificates from a recognized Certificate Authority and to develop an understanding how SSL certificates work. Real security is hard work. OPTIONS: -h Show this message -m Type of cert "client" or "server". Defaults to server. -f Config filename (openssl config file format) -c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed) -k CA key filename -C Common name (cert field) For a server cert, this should be the same address that clients attempt to connect to. Usually this will be the Fully Qualified Domain Name, but might be the IP of the server. For a CA or client cert, it is merely informational. Make sure your certs have unique common names. -O Org name (cert field) An informational string (company name) -o Output filename base (defaults to asterisk) -d Output directory (defaults to the current directory) Example: To create a CA and a server (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp This will create a CA cert and key as well as asterisk.pem and the the two files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will require you to import the ca.crt file as a trusted cert. To create a client cert using the CA cert created by the example above: ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user This will create client.crt/key/pem in /tmp. Use this if your device supports a client certificate. Make sure that you have the ca.crt file set up as a tlscafile in the necessary Asterisk configs. Make backups of all .key files in case you need them later. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	eeec27dafc	Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cb0d414f4b	Compile correctly on Linux (asterisk/localtime.h depends upon asterisk/autoconfig.h loading first). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	ce9d013cbd	Fix typo in SUSE init script. Reported by: Dave Cotton on asterisk-users list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	042d5edaa3	Fixes recursive lock problem in manager.c It is possible for a AMI session to freeze because of invalid use of recursive locks during the EVENT processing. This patch removes the unnecessary locks. (closes issue #18167) Reported by: sustav Patches: manager_locking_v1.diff uploaded by dvossel (license 671) Tested by: sustav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8c0627fada	Merged revisions 292556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292556 \| lmadsen \| 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) \| 6 lines Change res_ldap.sample.conf to match the schema. (closes issue #17376) Reported by: jcovert Patches: res_ldap.conf.sample.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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