20144 Commits (d348c9aa1ea5771b51ad9883420cb41a1fc27b3e)

 

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Author	SHA1	Message	Date
Leif Madsen	ea9186d4ea	Add 'soft hangup' alias per Steve Johnson on asterisk-users. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a8aef91e9d	Add example dialplan for dialing ISN numbers (http://www.freenum.org). ... Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	f7a34c978e	Add missing 'useragent' field to sip-friends.sql file. ... (closes issue #17171) Reported by: thehar Patches: sip-friends.patch uploaded by thehar (license 831) Tested by: pabelanger, thehar git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	31338f9671	Merged revisions 258029 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	910144937d	Update supported file extensions in doxygen. ... Updated the doxygen \arg line after looking at the file for some other Asterisk documentation and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	c7cf47ce7b	Change log message to match severity. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7965dd9509	Don't consider a missing indications.conf to be a critical error. ... There were many changes in revision 176627 which would avoid the error that a missing config would have caused. Other than this, there are no other config files (including asterisk.conf, surprisingly) that are required. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	990ccdd05f	Bad merge fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	38d77bde75	Blocked revisions 257856 via svnmerge ... ........ r257856 | jpeeler | 2010-04-19 14:09:46 -0500 (Mon, 19 Apr 2010) | 1 line make app_voicemail compile with IMAP_STORAGE ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6640f309a9	Commit compromise I suggested on review 608. ... This allows for multiple SRV queries to be done from the dialplan for the same service on a single call while still allowing one to bypass the call to SRVQUERY if they so please. Taking action since no comments had been left for a while. This can easily be reverted if needed. External tests still pass. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	9674766487	Fix incomplete CDR merge from r195881 ... Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3bb60ae5b7	Removing unused configuration parameters ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	77868073a8	Merged revisions 257686 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ba81ee6d28	Make sure to fail a monitor if we receive a negative response for a CC SUBSCRIBE. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	afedb856ae	Enable PRI SERVICE message support in chan_dahdi for the 'national' switchtype ... Revision 1072 of libpri added SERVICE message support for the 'national' switchtype. The attached patch enables the use of 'pri service' CLI commands on dahdi channels that are configured for the 'national' switchtype. (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch uploaded by dhubbard (license 733) Tested by: elguero, dhubbard Review: https://reviewboard.asterisk.org/r/612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8ced3317ed	Merged revisions 257544 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	84d0b95def	Merged revisions 257467 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	bef07ecc16	Merged revisions 257426 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	28b78c5e67	Merged revisions 257342 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7b73c324ac	Blocked revisions 257266 via svnmerge ... ........ r257266 | tilghman | 2010-04-14 18:08:11 -0500 (Wed, 14 Apr 2010) | 10 lines When forwarding a message, ensure that prepending works correctly. This is a regression in 1.4, only. (closes issue #17103) Reported by: mglazer Patches: 20100408__issue17103.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8b7a90a026	Yet another issue where the conversion of the application delimiter to comma caused an issue. ... Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e148ffc00f	Also unref the pvt when we delete the provisional keepalive job. ... (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2724f89bba	Merged revisions 257070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	012979b835	Ensure that we can have commas within cdr values. ... (closes issue #17001) Reported by: snuffy Patches: 20100412__issue17001.diff.txt uploaded by tilghman (license 14) Tested by: snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b1abf9234f	Update sample dialstrings in sip.conf.sample file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fb0a4e5bd0	Address Russell's comments on func_srv from reviewboard. ... * Change copyright date * Place channel in autoservice when doing SRV lookup * Get rid of trailing whitespace * Change logic in load_module function git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	69c252c290	Fix issue where recall would not happen when it should. ... Specifically, the situation would happen when multiple callers would request CC for a single generically-monitored device. If the monitored device became available but the caller did not answer the recall, then there was nothing that would poke the CC core to let it know that it should attempt to recall someone else instead. After careful consideration, I came to the conclusion that the only area of Asterisk that needed to be touched was the generic CC monitor. All other types of CC would require something outside of Asterisk to invoke a recall for a separate device. This was accomplished by changing the generic monitor destructor to poke other generic monitor instances if the device is currently available and the specific instance was currently not suspended. In order to not accidentally trigger recalls at bad times, the fit_for_recall flag was also added to the generic_monitor_instance_list struct. This gets set as soon as a monitored device becomes available. It gets cleared if a CCNR request triggers the creation of a new generic monitor instance. By doing this, we don't accidentally try to recall a device when the monitored device was being monitored for CCNR and never actually became available for recall in the first place. This error was discovered by Steve Pitts during in-house testing at Digium. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	47d8fdae97	Merged revisions 256900 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines Add How-To document on collecting debugging info for issues.asterisk.org Paul Belanger has been helping a lot with bug tracking recently and created this document that we can now point to when additional debugging information is required. This document will help those filing issues to know how to get the information required when filing their issues. This will make things easier on the developers. Initial text and changes by pabelanger. Tweaks and editing by myself. (closes issue #17159) Reported by: pabelanger Patches: HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	875014bdd4	Remove silly debug message that is not useful. ... (issue #17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bd53cbabcf	gives channel reference before unlocking it and using setvar helper. ... To guarantee the channel is valid when calling setvar on the MASTER_CHANNEL dialplan function, a channel reference must be taken before unlocking. Thanks to russell for pointing out the error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	d2e1f421fa	CLI command logger set level auto complete. ... A simple patch to enable auto tab complete. (closes issue #17152) Reported by: pabelanger Patches: 0017152.patch uploaded by pabelanger (license 224) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	387f2eb733	test_substitution expects func_curl to be present to work. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c78b29cb11	Add ASTERISK_FILE_VERSION() macro ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	16774318f9	fix hyphen vs. minus in man pages ... In troff '-' is used for a hyphen. A minus is denoted by '\-' . This is normally also used for a dash. This patch converts all '-'-s that are minuses or dashes to '\-'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9afa6af881	Remove status_response callbacks where they are not needed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	7509949658	Prevent crash when originating a call to a local channel. ... Call completion code tries to grab the call completion parameters from the requesting channel during local_request. When originating a call to a local channel, however, this channel is NULL. This was causing an issue for me when trying to run a test script. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d66b44b4ca	Merge CCSS architecture document from CCSS branch. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	537edff10f	Remove PRI CCSS BUGBUG message and update configure script. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	4b8f1c8cac	Add routines for parsing SIP URIs consistently. ... From the original issue report opened by Nick Lewis: Many sip headers in many sip methods contain the ABNF structure name-andor-addr = name-addr / addr-spec Examples include the to-header, from-header, contact-header, replyto-header At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success. I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure Nick has also included unit tests for verifying these routines as well, so...heck yeah. (closes issue #16708) Reported by: Nick_Lewis Patches: reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657 Review: https://reviewboard.asterisk.org/r/549 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ae7b76a1b9	Fix some compiler errors that popped up after the CCSS merge. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6cad0f1602	func_srv and explicit specification of a remote IP for SIP. ... From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0f01ace7af	Ensure that linker version scripts (used for symbol export control) always exist. ... Using wildcard matching in the Makefile is not adequate to determine whether an export file should exist for a module or not, so instead we'll just create one if the module needs one, or copy the default one if it does not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	bcbafc800e	Mac OS X does not support comparing a mutex to its initializer. Create a test for this. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d3f44c979c	fixes deadlock in chan_sip caused by usage of MASTER_CHANNEL dialplan function ... (closes issue #16767) Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/606/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1a2f96573a	Merged revisions 256225 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	3a4baef6d9	Fix for localchannel.tex to allow PDFs to be generated again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	5333a48b17	Using the Dial application f option when the call is forwarded will likely crash. ... Fix app_dial.c:do_forward() OPT_FORCECLID setting cid.cid_num with a stack allocated string instead of a heap allocated string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	a541609dde	Export MEETMEBOOKID and fix pin-less conferences with realtime conferences ... (closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago