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2012-09-12Skip any non-content information when looking for and handling content.Joshua Colp
This fixes a bug with Jitsi and conference calling. Jitsi implements XEP-0298 which places some conference-info information in the session-initiate request which chan_motif did not expect to occur. ........ Merged revisions 372995 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-12res_xmpp: Fix a segfault caused by bodyless messagesJonathan Rose
(closes issue ASTERISK-20361) Reported by: Noah Engelberth Review: https://reviewboard.asterisk.org/r/2108/ ........ Merged revisions 372984 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-12logger: Add rotatestrategy option of 'none' which does not perform rotationsJonathan Rose
With this option in use, it may be necessary to regulate your log files externally. (closes issue ASTERISK-20189) Reported by: Jaco Kroon Patches: asterisk-logger-norotate-trunk.patch uploaded by Jaco Kroon (license 5671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-12Add channel name to a warning to make debugging easier.Mark Michelson
The "autodestruct with owner in place" message is typically indicative of a channel reference leak. Printing out the name of the channel in the message may be helpful when trying to debug the issue. ........ Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372933 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372937 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-12Fixed r372696 when configured --disable-asteriskssl; properly install ↵David M. Lee
libasteriskssl.dylib on OS X. I didn't realize that libasteriskssl.c was still compiled, even when you disable asteriskssl; it simple gets statically linked into asterisk. ........ Merged revisions 372930 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-11chan_local: Switch from using a random 4 digit hex identifier to unique idJonathan Rose
Changes chan_local channels to use an 8 digit hex identifier generated atomically and sequentially in order to eliminate the chance of having multiple channels with the same name during high call volume situations. (issue ASTERISK-20318) Reported by: Dan Cropp Review: https://reviewboard.asterisk.org/r/2104/ ........ Merged revisions 372902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372916 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-11Fix inability to shutdown gracefully due to an unending channel reference.Mark Michelson
message.c makes use of a special message queue channel that exists in thread storage. This channel never goes away due to the fact that the taskprocessor used by message.c does not get shut down, meaning that it never ends the thread that stores the channel. This patch fixes the problem by shutting down the taskprocessor when Asterisk is shut down. In addition, the thread storage has a destructor that will release the channel reference when the taskprocessor is destroyed. (closes issue AST-937) Reported by Jason Parker Patches: AST-937.patch uploaded by Mark Michelson (License #5049) Tested by Jason Parker ........ Merged revisions 372885 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372888 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-11Fix bad channel application data reference.Mark Michelson
When channels get bridged due to an AMI bridge action or a DTMF attended transfer, the two channels that get bridged have their application data pointing to the other channel's name. This means that if one channel is hung up but the other moves on, it means that the channel that moves on will have its application data pointing at freed memory. (issue ASTERISK-20335) Reported by: aragon ........ Merged revisions 372840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372841 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372886 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-11Corrects the astsbindir setting when installing the sample asterisk.conf.David M. Lee
(closes issue ASTERISK-20406) ........ Merged revisions 372863 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372864 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-11chan_sip: Fix CHANGES and UPGRADE.txt for r372808Jonathan Rose
(issue AST-969) Reported by John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10chan_sip: Change SIPQualifyPeer to improve initial response timeJonathan Rose
Prior to this patch, The acknowledgement wasn't produced until after executing the sip_poke_peer action actually responsible for qualifying the peer. Now the response is given immediately once it is known that a peer will be qualified and a SIPqualifypeerdone event is issued when the process is finished. Thanks to OEJ for identifying the problem and helping to come up with a solution. (issue AST-969) Reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2098/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10Ensure iax2 debug output is displayed when expectedKinsey Moore
When IAX2 debug was changed from iax_showframe to iax_outputframe, some instances were missed (or added afterward). This was causing debug output to not be displayed when expected. (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by: John Covert ........ Merged revisions 372804 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372805 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372806 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10Deprecate chan_gtalk, chan_jingle, and res_jabberKinsey Moore
chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of using chan_motif and res_xmpp. They are a feature-equivalent replacement and are written to be more easily maintainable. (closes issue ASTERISK-20298) Review: https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen ........ Merged revisions 372795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10res_rtp_asterisk: Eliminate "type-punned pointer" build warning.David M. Lee
Removes "res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules" warning from the build on 32-bit platforms. The problem is that 'size' was referenced aliased to both (pj_size_t *) and (pj_ssize_t *). Now just make a copy of size that is the right type so there isn't any pointer aliasing happening. It also adds comments and asserts regarding what looks like an inappropriate use of pj_sock_sendto, but is actually totally fine. (closes issue ASTERISK-20368) Reported by: Shaun Ruffell Tested by: Michael L. Young Patches: 0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch uploaded by Shaun Ruffell (license 5417) slightly modified by David M. Lee. ........ Merged revisions 372777 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10app_meetme: Document that 'p' option will continue in dialplan.Jonathan Rose
(closes issue AST-991) Reported by John Bigelow ........ Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372767 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372768 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10Masquerade: Retain parkinglot settings made by CHANNEL function.Jonathan Rose
Prior to this patch, the user would have a parkinglot set on a channel that was parked and when the channel was retrieved, any attempt by that channel to park would simply use the default. This patch makes parkinglot values set in this way be retained through the masquerade. (closes issue AST-990) Reported by: Nick Huskinson Patches: masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 372736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372737 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372754 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-09Only re-create an SRTP session when neededMatthew Jordan
In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an SDP offer and the ability to re-create an SRTP session when the crypto keys changed. In certain circumstances - most notably when a phone is put on hold after having been bridged for a significant amount of time - the act of re-creating the SRTP session causes problems for certain models of phones. The patch committed in r356604 always re-created the SRTP session regardless of whether or not the cryptographic keys changed. Since this is technically not necessary, this patch modifies the behavior to only re-create the SRTP session if Asterisk detects that the remote key has changed. This allows models of phones that do not handle the SRTP session changing to continue to work, while also providing the behavior needed for those phones that do re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 ........ Merged revisions 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372710 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372711 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-08Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.David M. Lee
Without this flag, those files will compile with the system installed OpenSSL headers (if they exist). This is a real bummer if a different path was specified using --with-ssl= (closes issue ASTERISK-20392) ........ Merged revisions 372682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Recorded merge of revisions 372695 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Recorded merge of revisions 372696 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix MALLOC_DEBUG version of ast_strndup().Richard Mudgett
(closes issue ASTERISK-20349) Reported by: Brent Eagles ........ Merged revisions 372655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372656 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372657 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Remove annoying unconditional debug message from INC/DEC functions.Richard Mudgett
(closes issue AST-1001) Reported by: Guenther Kelleter ........ Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372629 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372630 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix exception path typo in app_queue.c try_calling().Richard Mudgett
(closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper ........ Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372625 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372626 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported ↵Richard Mudgett
values. The AMI action VoicemailUsersList VoicemailUserEntry event headers ServerEmail and MailCommand did not report the global values if they were not overridden. The VoicemailUserEntry event header ServerEmail was not populated with the global value if the voicemail user did not override it. The VoicemailUserEntry event header MailCommand was never populated with a value. * Removed unused struct ast_vm_user member mailcmd[]. (closes issue AST-973) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 372620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372621 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07svn:ignore cleanup.David M. Lee
* pjproject bin and lib directories should pretty much ignore everything * Ignore *.o in codecs/ilbc ........ Merged revisions 372611 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix parallel make for res_asterisk_rtp.David M. Lee
Fixes a build regression introduced in r369517 "Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1] http://svnview.digium.com/svn/asterisk?view=revision&revision=369517 When compiling asterisk in parallel like: $ make -j 10 It's possible to get errors like the following: .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep] Error 1 make[2]: *** [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2 make[3]: warning: jobserver unavailable: using -j1. Add `+' to parent make rule. This is because the build system is trying to build each of the libraries in pjproject in parallel. Now the build will build pjproject in a single job and link the results into res_asterisk_rtp. Parallel builds, on one test system, saves ~1.5 minutes from a default Asterisk build: Single job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys 0m15.970s Parallel make: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real 1m2.353s user 2m39.120s sys 0m18.850s (closes issue ASTERISK-20362) Reported by: Shaun Ruffel Patches: 0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch uploaded by Shaun Ruffel (License #5417) ........ Merged revisions 372609 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Free ast_str objects when temp file fails to be created in MiniVMMatthew Jordan
The previous commit (r372554) was from a patch that was written before r366880, which ensured that ast_str objects allocated in the sendmail routine were free'd in off nominal paths. This commit frees the string objects in the off nominal path introduced in r372554. (issue ASTERISK-17133) Reported by: Tzafrir Cohen ........ Merged revisions 372581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372582 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372583 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix file descriptor leak and pointer scope issue in MiniVM when sending mailMatthew Jordan
When MiniVM sends an e-mail and it has the volgain option set, it will spawn sox in a separate process to handle the manipulation of the sound file. In doing so, it creates a temporary file. There are two problems here: 1) The file descriptor returned from mkstemp is leaked 2) The finalfilename character pointer points to a buffer that loses scope once volgain processing is finished. Note that in r316265, Russell fixed some gcc warnings by using the return value of the mkstemp call. A warning was placed in minivm that the file descriptor was going to be leaked. This patch reverts that change, as it handles the leak and 'uses' the file descriptor returned from mkstemp. (closes issue ASTERISK-17133) Reported by: Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035) ........ Merged revisions 372554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372555 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372556 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Update QueueMemberStatus event documentation to include member status valuesMatthew Jordan
The Status: header in a QueueMemberStatus event (and other QueueMember* events) is the numeric value of the device state corresponding to that Queue Member. As those values are not exactly obvious, listing them in the documentation is useful. Matt Riddell reported this indirectly through the wiki page. (closes issue ASTERISK-20243) Reported by: Matt Riddell ........ Merged revisions 372531 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Fix loss of MOH on an ISDN channel when parking a call for the second time.Richard Mudgett
Using the AMI redirect action to take an ISDN call out of a parking lot causes the MOH state to get confused. The redirect action does not take the call off of hold. When the call is subsequently parked again, the call no longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames if it is already in a state where it is supposed to be sending MOH. The MOH may have been stopped by other means. (Such as killing the generator.) This simple fix is done rather than making the AMI redirect action post an AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus potentially breaking something with an unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches: jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 372521 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 372522 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372523 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Ensure listed queues are not offered for completionKinsey Moore
When using tab-completion for the list of queues on "queue reset stats" or "queue reload {all|members|parameters|rules}", the tab-completion listing for further queues erroneously listed queues that had already been added to the list. The tab-completion listing now only displays queues that are not already in the list. (closes issue AST-963) Reported-by: John Bigelow ........ Merged revisions 372517 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372518 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372519 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06chan_sip: Note change in behavior to how directmediapermit/deny ACL worksJonathan Rose
r366547 introduced a change to the directmedia ACL for chan_sip which modified the behavior significantly. Prior to the patch, this option would bridge peers with directmedia if a peer's IP address matched its own directmedia ACL. After that patch, the peer would check the bridged peer's ACL instead. This change has been present since 1.8.14.0. That patched failed to document the change in Upgrade.txt, so this patch adds mention of that change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876) ........ Merged revisions 372471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372472 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372473 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Ensure "rules" is tab-completable for "queue show"Kinsey Moore
Previously, tabbing at the end of "queue show" produced a list of available queues about which information could be shown, but did not include an alternative command, "rules", to access information about queue rules. The "rules" item should now be shown in the list of tab-completable items. (closes issue AST-958) Reported-by: John Bigelow ........ Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372445 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372446 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Fix DUNDi message routing bug when neighboring peer is unreachableMatthew Jordan
Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors, PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors. If the connection is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send messages to PBX3. If it does, PBX2 will assume that PBX3 already received the message and fail to forward the message on to PBX3 itself. This patch fixes this by only including peers in a DPDISCOVER message that are reachable by the sending node. This includes all peers with an empty address (00:00:00:00:00:00) and that are have been reached by a qualify message. This patch also prevents attempting to qualify a dynamic peer with an empty address until that peer registers. The patch uploaded by Peter was modified slightly for this commit. (closes issue ASTERISK-19309) Reported by: Peter Racz patches: dundi_routing.patch uploaded by Peter Racz (license 6290) ........ Merged revisions 372417 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372418 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372419 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Allow configured numbers for FollowMe to be greater than 90 charactersMatthew Jordan
When parsing a 'number' defined in followme.conf, FollowMe previously parsed the number in the configuration file into a buffer with a length of 90 characters. This can artificially limit some parallel dial scenarios. This patch allows for numbers of any length to be defined in the configuration file. Note that Clod Patry originally wrote a patch to fix this problem and received a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256 characters. Instead, the patch being committed duplicates the string in the config file on the stack before parsing it for consumption by the application. (closes issue ASTERISK-16879) Reported by: Clod Patry Tested by: mjordan patches: followme_no_limit.diff uploaded by Clod Patry (license #5138) Slightly modified for this commit. ........ Merged revisions 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372391 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372392 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Recorded merge of revisions 372373 from ↵Richard Mudgett
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix compile error. ........ Merged revisions 372372 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Correct documentation for ModuleLoad AMI actionKinsey Moore
The documentation incorrectly listed 'rtp' as a reloadable subsystem and left out many other reloadable subsystems. It is now also documented that subsystems may only be reloaded, not loaded or unloaded. (closes issue AST-977) Reported-by: John Bigelow ........ Merged revisions 372354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372358 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372365 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Ensure counts generated in manager_show_dialplan_helper are correctKinsey Moore
When manager_show_dialplan_helper was written, the counter increment for the total number of contexts was placed with the extensions increment instead of in the enclosing loop. This function should now generate correct context counts. (closes issue AST-970) Reported-by: John Bigelow ........ Merged revisions 372337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372338 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372340 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, ↵Alec L Davis
should be MF_GSIZE Remove unused goertzel_state_t member 'samples'. Related https://reviewboard.asterisk.org/r/2097/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Multiple revisions 372327-372328Richard Mudgett
........ r372327 | rmudgett | 2012-09-05 12:33:11 -0500 (Wed, 05 Sep 2012) | 15 lines Fix RTP/RTCP read error message confusion. The RTP/RTCP read error message can report "fail: success" when the read failure is because of an ICE failure. * Changed __rtp_recvfrom() to generate a PJ ICE message when ICE fails. * Changed RTP/RTCP read error message to indicate an unspecified error when errno is zero. (closes issue ASTERISK-20288) Reported by: Joern Krebs Patches: jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded by rmudgett (modified) ........ r372328 | rmudgett | 2012-09-05 12:35:20 -0500 (Wed, 05 Sep 2012) | 1 line Fix coding guidelines issue with a recent commit. ........ Merged revisions 372327-372328 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Re-fix sending unnegotiated payloads during a P2P RTP bridge.Mark Michelson
The previous fix still would look in the static_RTP_PT table, which is inappropriate since we specifically want to find a codec that has been negotiated. (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches: codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 372311 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Add fixes and cleanup to app_alarmreceiver.Mark Michelson
This work comes courtesy of Pedro Kiefer (License #6407) The work was posted to review board by Kaloyan Kovachev (License #5506) (closes issue ASTERISK-16668) Reported by Grant Crawshay (closes issue ASTERISK-16694) Reported by Fred van Lieshout (closes issue ASTERISK-18417) Reported by Kostas Liakakis (closes issue ASTERISK-19435) Reported by Deon George (closes issue ASTERISK-20157) Reported by Pedro Kiefer (closes issue ASTERISK-20158) Reported by Pedro Kiefer (closes issue ASTERISK-20224) Reported by Pedro Kiefer Review: https://reviewboard.asterisk.org/r/2075 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Fix memory leaks in app_voicemail when using IMAP storage or realtime configMatthew Jordan
This patch fixes two memory leaks: 1. When find_user is called with NULL as its first parameter, the voicemail user returned is allocated on the heap. The inboxcount2 function uses find_user in such a fashion when counting new messages, and fails to free the resulting voicemail user object. 2. When populate_defaults is called on a voicemail user, it wipes whatever flags have been set on the object by copying over the global flags object. If the VM_ALLOCED flag was ste on the voicemail user prior to doing so, that flag is removed. This leaks the voicemail user when free_user is later called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) Patch slightly modified for this commit. Review: https://reviewboard.asterisk.org/r/2096 ........ Merged revisions 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372288 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372289 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05LDAP Realtime Peers Cannot RegisterDarren Sessions
Prior to 1.8, it was not necessary for an explicit "type" to be set for an asterisk LDAP realtime peer. Now the routine find_peer actually checks the type field during registration and fails to find the peer if it is not set. The attached patch makes the realtime type equal whatever type is being searched for if the type is 0 upon return from routine build_peer. (closes issue ASTERISK-17222) Reported by: John Covert Patch by: David Vossel Tested by: Darren Sessions Review: https://reviewboard.asterisk.org/r/2095/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Fix breakage caused by last merge. Missing a variable for 11 and trunk.Michael L. Young
........ Merged revisions 372266 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05dsp.c: Fix multiple issues when no-interdigit delay is present, and fast ↵Alec L Davis
DTMF 50ms/50ms Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary; 1. reseting of hits=0, when no signal, only need to set it once. 2. incrementing of hits, when the hit is the same as the current hit. 3. setting of lasthit, when it's the same as before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3 spelling mistakes (closes issue ASTERISK-19610) alecdavis (license 585) Reported by: Jean-Philippe Lord Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2085/ ........ Merged revisions 372239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372240 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372241 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and ↵Alec L Davis
tone_detect use a temporary short int when repeatedly used to call goertzel_sample. alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2093/ ........ Merged revisions 372212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372213 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372214 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Fix Incrementing Sequence Number For Retransmitted DTMF End PacketsMichael L. Young
In Asterisk 1.4+, a fix was put in place to increment the sequence number for retransmitted DTMF end packets. With the introduction of the RTP engine API in 1.8, the sequence number was no longer being incremented. This patch fixes this regression as well as cleans up a few lines that were not doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh Bansal Tested by: Michael L. Young Patches: 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2083/ ........ Merged revisions 372185 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372198 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372199 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Fix memory leak when CEL is successfully written to PostgreSQL databaseMatthew Jordan
PQClear is not called when the result object of a call to PQExec has a status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was handled properly, so this memory leak only occurred when CEL records were successfully written. This patch properly clears the result in the nominal code path. (closes issue ASTERISK-19991) Reported by: Etienne Lessard Tested by: Etienne Lessard patches: mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394) ........ Merged revisions 372158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372165 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372175 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04app_queue: PAUSEALL/UNPAUSEALL logged only if interface is a queue memberJonathan Rose
Adding UPGRADE.txt entry for r372148 (issue AST-946) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04app_queue: Only log PAUSEALL/UNPAUSEALL when 1+ memebers changed.Jonathan Rose
Prior to this patch, if pause or unpause was issued on an interface without specifying a specific queue, a PAUSEALL or UNPAUSEALL event would be logged in the queue log even if that interface wasn't a member of any queues. This patch changes it so that these events are only logged when at least one member of any queue exists for that interface. (closes issue AST-946) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2079/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04Fix issue where SIP devices were not notified when custom devices changed to ↵Mark Michelson
"ringing". The problem had to do with logic used when checking for what the oldest ringing channel was. The problem was that if no channel was found, then no notification would be sent. For custom device states, there is no associated channel, so no notification would get sent. This fixes the issue by still sending the notification even if no associated channel can be found for a ringing device state change. (closes issue ASTERISK-20297) Reported by Noah Engelberth ........ Merged revisions 372137 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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