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2010-06-18file.c was truncating audio file formats to the lower 32bits.dvossel1-6/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271341 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17adds support for slin16 in sipdvossel1-3/+5
(closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17adds speex 16khz audio supportdvossel3-11/+16
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Set sin_family to AF_INET when doing lookups, also reset sin_port the first ↵mnicholson2-9/+12
time the ip address changes. (closes issue #17496) Reported by: ManChicken (closes issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch uploaded by chappell (license 8) Tested by: DennisD, gentlec, damage, wimpy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270974 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of G.719 pass-through supportdvossel3-0/+12
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Don't continue sending the file when there has been an errortwilson1-0/+1
If there is a problem with a firmware file, Polycom phones will close the connection. We were continuing to send the file anyway. There should be no reason to continue sending a file if there is an error writing it. (closes issue #16682) Reported by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270692 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Merged revisions 270583 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines Variables have always been case-sensitive, so we should not be removing case-insensitive matches. Bug reported via the -dev list. See http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270584 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11Merged revisions 269960 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines For SpeeX, 0 bits remaining is valid and does not need an emitted warning. (closes issue #15762) Reported by: nblasgen Patches: issue15672.patch uploaded by pabelanger (license 224) Tested by: nblasgen ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269976 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11Add DBGetComplete event after a DBGetResponse.tilghman1-0/+4
(closes issue #16965) Reported by: rrb3942 Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269938 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11Remove lines from the output related to the backtrace itself.tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269936 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Merged revisions 269821 via svnmerge from mmichelson1-4/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines Fix potential crash when writing raw SLIN audio on a PLC-enabled channel. The issue here was that the frame created when adjusting for PLC had no offset to its audio data. If this frame were translated to another format prior to being sent out an RTP socket, all went well because the translation code would put an appropriate offset into the frame. However, if the SLIN audio were not translated before being sent out the RTP socket, bad things would happen. Specifically, the ast_rtp_raw_write makes the assumption that the frame has at least enough of an offset that it can accommodate an RTP header. This was not the case. As such, data was being written prior to the allocation, likely corrupting the data the memory allocator had written. Thus when the time came to free the data, all hell broke loose. ....Well, Asterisk crashed at least. The fix was just what one would expect. Offset the data in the frame by a reasonable amount. The method I used is a bit odd since the data in the frame is 16 bit integers and not bytes. I left a big ol' comment about it. This can be improved on if someone is interested. I was more interested in getting the crash resolved. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Ensure that 'logger show channels' works properly when wildcards are used in ↵kpfleming1-1/+1
logger.conf. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269707 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Merged revisions 269635 via svnmerge from tilghman2-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines Ensure restartable system calls can restart (BSD signal semantics). This eliminates the annoying <beep> on the console. (closes issue #17477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269636 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Attempt to fix FreeBSD build problem.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Resolve an invalid memory read on an event.russell1-4/+13
Valgrind pointed out that attempting to get an IE value from an event that has no IEs produces an invalid memory read past the end of the event. Thanks to mmichelson for pointing the problem out to me and then testing the fix. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269334 via svnmerge from pabelanger1-6/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines Fix Debian init script to not use -c. When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. We now handle color displays properly. (closes issue #16784) Reported by: pabelanger Patches: 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger, tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269346 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Fix some doxygen warnings.lmadsen5-7/+8
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add SRTP support for Asterisktwilson4-0/+126
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ↵tilghman1-1/+1
is expressed as a double, the last value would not also need to be a double. (closes issue #15807) Reported by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268773 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Event well was going dry.tilghman1-0/+5
(issue #17234) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268731 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Set threshold for silence detection defaults to 256pabelanger1-5/+17
(closes issue #15685) Reported by: david_s5 Patches: dsp-silence-threshold-init.diff uploaded by dant (license 670) issue15685.patch.v5 uploaded by pabelanger (license 224) Tested by: danti Review: https://reviewboard.asterisk.org/r/670/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268690 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Suppress warning in waitstream_core().rmudgett1-0/+3
Suppress the warning about unexpected control subclass frames for AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC in file.c:waitstream_core(). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268578 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Fix crash in DTMF detection.tilghman1-9/+4
What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue #17371) Reported by: alecdavis (closes issue #17474) Reported by: kenner git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Verify event is not NULL before attempting to lower its usecount.tilghman1-1/+3
(closes issue #17234) Reported by: mav3rick git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268454 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Remove a LOG_WARNING.russell1-1/+0
This came up when using the sample configs, and just indicates expected behavior. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267714 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Remove unnecessary code relating to PLC.mmichelson1-40/+5
The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. Review: https://reviewboard.asterisk.org/r/683/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267492 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Malicious Call ID support.rmudgett1-18/+2
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267350 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been ↵russell1-1/+1
specified. When ASTCFLAGS was specified with the make command, Makefile.rules was using the specified value from the command line and not the one here, making it so this flag would go missing. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267303 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add a CLI command that blocks until Asterisk has fully booted.russell1-0/+24
Review: https://reviewboard.asterisk.org/r/684/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267138 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett5-0/+1954
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267009 via svnmerge from pabelanger1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun 2010) | 7 lines Cleanup error/warning messages in AEL2 parser (closes issue #16684) Reported by: Silmaril Patches: patch_ael2_logmsg.diff uploaded by Silmaril (license 979) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267041 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.rmudgett1-0/+1
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267008 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02pthread_join to assure the thread is really gonepabelanger1-1/+6
(closes issue #15465) Reported by: fnordian Patches: bridging.patch uploaded by fnordian (license 110) Tested by: lmadsen, fnordian, peterh Review: https://reviewboard.asterisk.org/r/679/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266877 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵tilghman1-8/+129
pager messages). (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Eliminate stale manager events after a set interval, even if AMI clients ↵tilghman1-40/+52
don't query for them. Actions (or failures to act) by external clients should not cause memory leaks in Asterisk, especially when those continued leaks could cause Asterisk to misbehave later. (closes issue #17234) Reported by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by tilghman (license 14) 20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14) Tested by: mav3rick, davidw (closes issue #17365) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266682 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Merged revisions 266585 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines Prevent CLI prompt from distorting output of lines shorter than the prompt. Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266592 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Setup environment variables for the benefit of child processes and disallow ↵tilghman1-0/+13
changing them. (closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266385 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Only report swap on platforms which can examine those statisticstilghman1-2/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266337 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Merged revisions 266142 via svnmerge from tilghman2-22/+44
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266146 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Fix misspelling of macro args.mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266092 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26do all sip registry parsing before transmit_registerdvossel1-1/+5
This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Memory leak in connected line data when SIP blond transfer done.rmudgett1-23/+11
The handling of the control subclass AST_CONTROL_READ_ACTION frame leaked connected line string memory in __ast_read(). Also in __ast_read() the frame type switch should not have had a case for AST_CONTROL_READ_ACTION. AST_CONTROL_READ_ACTION is not a frame type. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265608 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merge the rest of the FullyBooted patchtwilson2-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265467 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24On systems with a LOT of RAM, a signed integer sometimes printed negative.tilghman1-10/+11
(closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265316 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24fixes segfault when using generic plcdvossel1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265273 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Channel initialization failure causes crashes.rmudgett1-74/+85
__ast_channel_alloc_ap() has several points in the initialization of a new channel structure where it could fail. Since the channel structure is now an ao2 object, the destructor callback needs to be able to handle clean up when the structure setup is incomplete. Problems corrected: 1) Failing to setup the alertpipe would not unreference the structure but free it directly. Doing this to an ao2_object is very bad. 2) File descriptors need to be initialized to -1 before a construction failure could occur so the destructor will not close unopened descriptors. 3) The destructor needs to check that the string field has been initialized before using any string field values. Crashes expected. 4) The destructor should not notify devstate if the device name is empty. It is a waste of cycles and a couple ERROR log messages are generated. Review: https://reviewboard.asterisk.org/r/675/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265174 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 264996 via svnmerge from mmichelson2-27/+57
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264997 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Merged revisions 264820 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines ast_callerid_parse() had a path that left name uninitialized. Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Let ExtensionState resolve dynamic hints.tilghman1-0/+11
(closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264779 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Avoid crash in generic CC agent init if caller name or number is NULL.rmudgett1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264711 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-17 01:30:02 +0000