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2009-01-14Merged revisions 168593 via svnmerge from twilson1-4/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168594 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Clarify a message that app_queue prints and change to a debug-level messagemmichelson1-1/+1
The "No one is answering..." verbose message contained 3 numbers that were not explained in any way to whoever was viewing the message. It is more helpful now since the message explains what the numbers mean. Also, the message has been downgraded to "DEBUG" level. (closes issue #14172) Reported by: caio1982 Patches: queue_answering_debug.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168579 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Merged revisions 168561 via svnmerge from russell3-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168562 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12Better to use the proper app nameoej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168497 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-09Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentationtwilson1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167973 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-09Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be settwilson1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167935 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Fix variables to comply with documentation changestilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167837 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Textual changes, consistency in status variable naming, and other minor bugs.tilghman1-60/+65
(closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167835 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Add the average talk time for a queuemmichelson1-11/+19
This patch adds the functionality to app_queue of calculating the average amount of time that channels are bridged for a queue. The algorithm used to calculate the average is the same exponential average currently used to calculate the average holdtime. See the CHANGES file to see the methods you may use to view this information. (closes issue #13960) Reported by: coolmig Patches: app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167792 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-07 Answer the channel if it has not already been answered and we've already ↵bweschke1-0/+5
found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167478 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-29Update app_queue to deal with the removal of AST_PBX_KEEPALIVEmmichelson2-5/+7
When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166861 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-23Merged revisions 166093 via svnmerge from murf3-53/+33
https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166665 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Fix the XML documentation for Record().eliel1-6/+6
<value> tags inside <variable> elements must have CDATA and no another XML node. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165954 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Merged revisions 165889 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165890 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165767 via svnmerge from tilghman1-10/+52
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165797 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Numerous documentation updates.file1-8/+8
(closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165792 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.russell1-2/+6
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165723 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Fix 2 resource leaks and fix another pipe-to-comma conversiontilghman1-5/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165658 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Add a \todo note for app_originate.russell1-0/+4
Jared Smith suggested that we add a way to be able to set variables and functions on the outbound channel. I think that it's a great idea, so I have added it as a todo so that it gets done at some point. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165469 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Add a new application, Originate.russell1-0/+213
(closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165433 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Add RECORD_STATUS variable, as requested on the -users list.tilghman1-0/+22
Patch by me (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165397 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Fix the buildmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165326 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Oops, broke trunktilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165325 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165317 via svnmerge from tilghman1-21/+55
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165319 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165255 via svnmerge from mmichelson4-3/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165318 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17And actually assign the function to a pointer...mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165143 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Use the create_vm_state_from_user function in a place wheremmichelson1-18/+7
it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165142 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16(closes issue #13669)jpeeler1-0/+1
Reported by: pj Delete file recording if recording terminated from a hangup. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164942 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Merged revisions 164876 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has been returned. This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164877 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was ↵russell1-4/+6
not needed. (closes issue #14081) Reported by: pkempgen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164623 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Add an 'i' option to app_page. This option works the same asmmichelson1-0/+9
the 'i' options for app_dial and app_queue, in that they will ignore any attempts by phones to forward the call. (closes issue #13977) Reported by: putnopvut Patches: page_ignore_forwards.patch uploaded by putnopvut (license 60) Tested by: putnopvut, acunningham git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164428 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Fix a compile warning and a logic error that could have been badmmichelson1-2/+2
for non-realtime queues git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Fix up a few issues with regards to queuesmmichelson1-12/+23
* Fix reference counting used in the __queues_show function * Add code to be sure that the "queue show" command does not print information for a realtime queue which has been deleted from the backend * Add a missing unref to the realtime queue loading function for the case where a queue is in the module's container but has been deleted from the realtime backend (closes issue #14033) Reported by: cristiandimache Patches: 14033.patch uploaded by putnopvut (license 60) Tested by: cristiandimache git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Make app_fax compatible with newer versions of spandsp. This remains ↵file1-0/+3
backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164257 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Fix build WRT ast_str_opaquerussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164202 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)tilghman7-43/+42
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163991 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Only detach and destroy the whisper audiohooks if they are actually in use.file1-8/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163912 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-12When using realtime queues, app_queue wasn't updating the strategy if it was ↵twilson1-2/+20
changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163873 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Add an option to voicemail.conf to allow urgent messages to bemmichelson1-1/+10
forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163213 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163084 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163080 via svnmerge from mmichelson1-21/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163081 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-10Finish conversion to using ARRAY_LEN and remove it as a janitor project.file1-1/+1
(closes issue #14032) Reported by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162542 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162463 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162466 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162348 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162355 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162341 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162342 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162286 via svnmerge from russell1-7/+27
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162291 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162273 via svnmerge from file1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162275 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162014 via svnmerge from russell1-2/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162016 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-08Add voicemail related applications and functions XML documentation:eliel1-80/+185
applications: - VoiceMail() - VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - MAILBOX_EXISTS() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161604 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-07Introduce SMS() application XML documentation.eliel1-28/+50
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161571 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-26 15:08:02 +0000