22296 Commits (0e1ac09a2bc062ef627b1e07057cfb49e0c8d24c)

 

All Branches   

Search

Author	SHA1	Message	Date
Automerge script	0e1ac09a2b	Merged revisions 370954 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370954 | elguero | 2012-08-08 17:42:05 -0500 (Wed, 08 Aug 2012) | 26 lines Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9dfb7700c4	Merged revisions 370924 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370924 | kmoore | 2012-08-08 15:29:16 -0500 (Wed, 08 Aug 2012) | 9 lines Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	105b05218a	Merged revisions 370901 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370901 | rmudgett | 2012-08-08 15:04:44 -0500 (Wed, 08 Aug 2012) | 5 lines Fix the analog dial *0 flash-hook of bridged peer feature. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	cb11c1cf4d	Merged revisions 370858 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370858 | kmoore | 2012-08-07 14:21:54 -0500 (Tue, 07 Aug 2012) | 5 lines Add missing AST_CAUSE_* -> text translations ........ Merged revisions 370856 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	64bf23b2d2	Merged revisions 370798 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370798 | mmichelson | 2012-08-06 10:02:04 -0500 (Mon, 06 Aug 2012) | 7 lines Improve debug message for temporary outbound proxies. Thanks to Paul Belanger for pointing this out. ........ Merged revisions 370797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	adc08df2d1	Merged revisions 370772 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370772 | mmichelson | 2012-08-03 16:50:29 -0500 (Fri, 03 Aug 2012) | 41 lines Multiple revisions 370769-370771 ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 ........ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines Remove unused variable. ........ r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	81fdb9dfc9	Merged revisions 370698 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370698 | kmoore | 2012-07-31 21:26:09 -0500 (Tue, 31 Jul 2012) | 8 lines Revert alloca changes for utils These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". ........ Merged revisions 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	764c114bee	Merged revisions 370672 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370672 | mjordan | 2012-07-31 16:19:41 -0500 (Tue, 31 Jul 2012) | 24 lines Schedule pokes of registered SIP peers within a given timespan after SIP reload With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 ........ Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	86776cb4d5	Merged revisions 370643 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370643 | kmoore | 2012-07-31 14:57:09 -0500 (Tue, 31 Jul 2012) | 12 lines Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	26577d1d93	Merged revisions 370619 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370619 | mmichelson | 2012-07-31 10:31:57 -0500 (Tue, 31 Jul 2012) | 34 lines Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan ........ Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b5a26ce601	Merged revisions 370564 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500 (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on error path in chan_misdn. ........ Merged revisions 370563 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ec96eda533	Merged revisions 370547 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) | 5 lines app_meetme: Change app_meetme support level to extended from deprecated (closes issue ASTERISK-20134) Reported by: Leif Madsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	54f01be015	Merged revisions 370495 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370495 | jrose | 2012-07-25 16:12:50 -0500 (Wed, 25 Jul 2012) | 14 lines res_agi: Add message indicating need for \n character in verbose message The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper ........ Merged revisions 370494 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e3f9e1b64d	Revert a change that broke compilation ... 1) There is no such function as ast_ref() 2) The patch was originally credited as the one uploaded by Guenther Kelleter (license 6372) via issue AST-921, but the patch committed was not the patch referenced on the issue. 3) Guenther Kelleter's patch was actually correct. It moved the ast_free above the presencechange_cleanup label. I am not committing his change as it is not technically necesary--calling ast_free(NULL) is perfectly safe and I worry that moving the ast_free outside of the label could lead to future bugs if someone ever adds another failure conditional and expects 'goto presencechange_cleanup;' to clean up after everything. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ff8e14d782	Don't attempt free of NULL ptr in pbx.c handle_presencechange ... (closes issue AST-921) Reported by: Guenther Kelleter Patches: nullptr.patch uploaded by Guenther Kelleter (license 6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a2fe225529	Merged revisions 370430,370432 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500 (Tue, 24 Jul 2012) | 5 lines Rewrite a comment that didn't adequately explain the code it was documenting. ........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500 (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error message Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	33cbee2d22	Merged revisions 370384 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500 (Mon, 23 Jul 2012) | 5 lines Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2dec8e2a0a	Merged revisions 370361 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500 (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to dummy channels. Revision 370205 added the use of a datastore attached to a dummy channel to resolve a memory leak, but ast_dummy_channel_destructor() in this branch did not free datastores, resulting in a continued (but slightly smaller) memory leak. This patch backports the change to free said datastores from the Asterisk trunk. (related to issue AST-916) ........ Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	5912f75512	Merged revisions 370271,370274,370277 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500 (Thu, 19 Jul 2012) | 49 lines Handle extremely out of order RFC 2833 DTMF The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (issue ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt Jordan ........ Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500 (Thu, 19 Jul 2012) | 17 lines Fix compilation error when MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500 (Thu, 19 Jul 2012) | 7 lines Fix compiler warnings. gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1440b3f1ec	Merged revisions 370206 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500 (Wed, 18 Jul 2012) | 19 lines Resolve severe memory leak in CEL logging modules. A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	8d9bfd3342	Merged revisions 370184 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500 (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b2ab0e1d0c	Merged revisions 370132 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500 (Mon, 16 Jul 2012) | 11 lines Code cleanup and bugfix in chan_sip outboundproxy parsing. The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a0fec72b8b	Merged revisions 370082 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500 (Mon, 16 Jul 2012) | 8 lines Add comments about the BUILD_NATIVE change This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2190019e49	Merged revisions 370015,370025 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500 (Thu, 12 Jul 2012) | 11 lines Include Expires header for SIP PUBLISH requests RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500 (Thu, 12 Jul 2012) | 8 lines Add missing ast_hangup() calls on some analog exception paths. Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	825c3eaf2b	Merged revisions 369994 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500 (Thu, 12 Jul 2012) | 12 lines Prevent double uri_escaping in chan_sip when pedantic is enabled If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported-by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2e1130db73	Merged revisions 369971 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369971 | elguero | 2012-07-12 09:25:45 -0500 (Thu, 12 Jul 2012) | 14 lines Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f59ef2b922	Merged revisions 369938 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500 (Wed, 11 Jul 2012) | 11 lines Allow the REALTIME() function to report errors back to the caller. Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	09a61e14d1	Merged revisions 369871 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500 (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related documentation Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	890f7bf9e7	Add support for exposing the received contact URI and also for setting the request URI in messages. ... (closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ead4555a90	Merged revisions 369819 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369819 | qwell | 2012-07-09 12:06:40 -0500 (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f6c99d233e	Merged revisions 369793 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369793 | jrose | 2012-07-09 09:43:49 -0500 (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral change accidentally introduced in r369750 When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d8930ab6d1	Merged revisions 369751 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369751 | jrose | 2012-07-06 16:02:37 -0500 (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH control frames so that we don't display a warning. chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	19afbc2af2	Merged revisions 369732 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500 (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous freeing of an SSL_CTX. The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b4dcb67e17	Merged revisions 369709 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500 (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f4c2e137bb	Merged revisions 369653 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500 (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with voicemail The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1bf80b527c	Merged revisions 369627 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500 (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a provisional response arrives during a re-INVITE Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f477a9e831	Merged revisions 369580 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369580 | twilson | 2012-07-03 12:02:18 -0500 (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs timing out after a provisional response There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ad64385217	Merged revisions 369558 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369558 | twilson | 2012-07-03 09:34:22 -0500 (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4fb96676bd	Merged revisions 369511 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun 2012) | 3 lines Fix apparent copy and paste error where incorrect "glue" is used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c2711a357e	Merged revisions 369491 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri, 29 Jun 2012) | 5 lines With some configurations a transport is not actually specified so assume UDP in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f51af261fe	Merged revisions 369472 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri, 29 Jun 2012) | 10 lines Make the address family filter specific to the transport. (closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	23db01bcc8	Merged revisions 369437 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369437 | twilson | 2012-06-27 16:10:01 -0500 (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. (closes issue ASTERISK-19992) ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3fe87f3c47	Merged revisions 369391 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500 (Tue, 26 Jun 2012) | 15 lines Fix crash in unloading of res_adsi module When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. ........ Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4478e3580b	Merged revisions 369369 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500 (Mon, 25 Jun 2012) | 29 lines Fix incorrect duration reporting in CDRs created in batch mode Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ ........ Merged revisions 369351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b25a80690c	Merged revisions 369353 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500 (Mon, 25 Jun 2012) | 14 lines Re-fix how local tag is generated when sending a 481 to an INVITE. Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	108f718fd5	Merged revisions 369325,369328 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500 (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324 ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module. The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 ........ r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines Forgot to svn add this file in my last commit. ........ Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500 (Mon, 25 Jun 2012) | 15 lines Fix Bridge application occasionally returning to the wrong location. * Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque ........ Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	03b52a9b84	Merged revisions 369303 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500 (Mon, 25 Jun 2012) | 14 lines Be more consistent with the return code for requests received from invalid domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	17a5e814f5	Merged revisions 369283 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500 (Fri, 22 Jun 2012) | 22 lines Fix Bridge application and AMI Bridge action error handling. * Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. ........ Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f3cc355870	Merged revisions 369259,369263 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F and F(x) action logic in Dial application. ........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500 (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in app Queue rather than a polluted res2 value. ........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9c288ea612	Merged revisions 369236,369239 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500 (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie hangup debug message. They are all zombies now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic CCSS recall. ........ Merged revisions 369238 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago