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2008-11-07If 'asterisk.conf' is not found, instead of giving up,eliel1-10/+7
load documentation for the 'en_US' language (fix my last commit). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155204 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-07Fix an asterisk crash if no asterisk.conf configuration file is present.eliel1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155175 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-06Simplify the output of [See Also].eliel1-10/+9
Functions are printed without parenthesis like: FUNTION Applications are printed with parenthesis like: AppName() Cli commands are printed like: 'core show application' The other type of references are printed as they are inside the <ref> tag. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154967 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Update a couple places to use the new ast_channel_search_locked API call.seanbright1-13/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154923 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Don't read history on -rx commands.tilghman1-12/+14
(Closes issue #13571) Reported by: tzafrir Patch '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by tzafrir. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154922 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Add LISTFILTER dialplan function, along with supporting documentation. Seetilghman1-0/+13
documentation for more information on how to use it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154915 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Merged revisions 154685 via svnmerge from murf1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154687 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Add more SeeAlso references based on TFOT.eliel2-3/+111
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05- Add more <see-also> based on TFOT.eliel1-0/+2
- Add the 'filename' type to the see-also ref. To be able to reference a filename. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154578 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04Introduce a new API call ast_channel_search_locked, which iterates through theseanbright1-0/+18
channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154429 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04Slightly optimize ast_devstate_str and rename global functions devstate2str ↵tilghman4-49/+30
and config_text_file_save to have an ast_ prefix git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154260 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04GLOB_BRACE is already added to MY_GLOB_FLAGS if it is supported on theseanbright1-1/+1
platform. This should resolve some build errors on Solaris. (issue #13704) Reported by: dougm git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154191 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04Fix build errors.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154186 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-03Merged revisions 154060 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154061 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 ↵kpfleming9-95/+194
branch, and add the ones needed for all the new code here too git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153616 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell7-465/+2789
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsmmichelson1-2/+2
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Recent CDR fixes moved execution of the 'h' exten into the bridging code, so ↵twilson1-0/+7
variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153181 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Use the ast_str API call to reset the string instead of manually editing its ↵russell1-2/+1
internals (closes issue #13816) Reported by: eliel Patches: channel.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153057 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30Merged revisions 152811 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct 2008) | 3 lines instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152812 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30fix a few small things found by using sparsekpfleming4-17/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152809 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30After seeing another problem in #asterisk stemming frommmichelson1-1/+1
the low default value of featuredigittimeout, I decided it was high time to change it. I have changed the default to 2000 ms based on a suggestion from Leif Madsen. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152807 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30Track down and fix annoying lock errorstilghman1-5/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152689 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29Merged revisions 152535 via svnmerge from murf1-60/+89
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152536 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-25Merged revisions 151905 via svnmerge from russell1-5/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines Move AMI initialization to occur after loading modules. This prevents a deadlock when someone tries to initiate a module reload from the AMI just as Asterisk is starting. (closes issue #13778) Reported by: hotsblanc Fix suggested by hotsblanc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@151906 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-20 Do NOT attempt to do anything with the ast_config struct when it's been ↵bweschke1-2/+2
returned as INVALID by the config file interpreter. (closes issue #13741) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@151246 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-19cleaup of the TCP/TLS socket API:kpfleming3-161/+176
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: http://svn.digium.com/svn/asterisk/trunk@151101 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-18 Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) ↵bweschke1-2/+9
when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150817 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Fix CLI command 'channel request hangup'mvanbaak1-2/+2
Prodded on IRC by Russell and fixed by eliel (closes issue #13730) Reported by: eliel Patches: main_cli.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150664 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-16Merged revisions 150304 via svnmerge from mmichelson1-4/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions *always* required that a channel be present. I need to go home earlier, I think :) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150305 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-16Merged revisions 150298,150301 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto ........ r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines And don't forget to return on the error condition ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150302 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merged revisions 149204 via svnmerge from mmichelson1-2/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149205 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Add additional memory debugging to several core APIs, and fix several memorytilghman3-3/+37
leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149199 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merged revisions 148611 via svnmerge from kpfleming1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines it would be nice if this message printing code had actually been tested before it was committed... ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148612 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merge realtime_update2 branch, which adds a new realtime API call namedtilghman1-15/+32
'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148570 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-13Hmmm. Nobody (but me) is interested in seeing murf1-10/+106
the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148519 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-13Highlightning even more bugs in the current tcp/tls implementation.oej1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148473 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-10Don't include logger.h in asterisk.h by default as it is causing problems ↵seanbright2-0/+2
building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148200 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09The priority was unnecessary for the manager atxfer, so it hasmmichelson1-15/+4
been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148160 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Merged revisions 146026 via svnmerge from mmichelson1-4/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148112 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Reverting format addition for nowtilghman1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148071 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Fudges for wav16, just like wav49tilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148070 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09(closes issue #13139)jpeeler1-0/+4
Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147952 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09(closes issue #13557)murf6-8/+8
Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147807 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-07Allow people to select the old console behavior of white text on a blacktilghman2-4/+37
background, by using the startup flag '-B'. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147262 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-07Explicitly setting these fields to NULL was done because I wasn't sure if ↵jpeeler1-3/+0
they would be NULL otherwise. Since they will be set automatically, removing. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147146 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06Similar to r143204, masquerade the channel in the case of Park being called ↵jpeeler1-1/+1
from AGI. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146923 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06This commit squashes together three commits because the wrong approach was ↵jpeeler1-17/+40
originally used. (One of the commits was only one line.) 1) r143204: The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel. 2) r143270: Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced. 3) r143475: (the one liner) compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146883 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06fix some comment placementjpeeler1-2/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146877 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06Explicitly set args in park_call_exec NULL so in the case of no options ↵jpeeler1-12/+20
being passed in, there is no garbage attempted to be used. Also, do not set args to unknown value again if there are no options passed in. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146875 f38db490-d61c-443f-a65b-d21fe96a405b
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