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2008-01-09Added a new module, res_phoneprov, which allows auto-provisioning of phonestwilson2-5/+115
based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97622 via svnmerge from qwell1-5/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a command could not be found. Also fix a comment which may have led to this happening. Issue 11718, reported by kshumard. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97623 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97618 via svnmerge from qwell1-3/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line Fix some locking and return value funkiness. We really shouldn't be unlocking this lock inside of a function, unless we locked it there too. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97620 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09New option in trunk, needs strdupa to be safe, tootilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97350 via svnmerge from tilghman2-47/+45
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) | 5 lines Allow filename completion on zero-length modules, remove a memory leak, remove a file descriptor leak, and make filename completion thread-safe. Patched and tested by tilghman. (Closes issue #11681) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97364 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Merged revisions 97194 via svnmerge from tilghman2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines Increase constants to where we're less likely to hit them while debugging. (Closes issue #11694) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97198 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Merged revisions 97077 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines Apply multiple crash fixes, found in issue #11386, but not completely closing that issue. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97125 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Display a message if no config mappings are found with "core show config ↵qwell1-8/+13
mappings". Closes issue #11704, patch by kshumard. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96936 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. ↵file2-107/+100
Previously they would get registered twice because of the way manager.c operates. (closes issue #11699) Reported by: caio1982 Patches: manager_module_commands1.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96858 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Now that the version.h file was getting properly regenerated every time the svnrussell4-11/+11
revision changed, every module that used the version was getting rebuilt after every svn update. This severly annoyed me pretty quickly, so I have improved the situation. Now, instead of generating version.h, main/version.c is generated. version.c includes the version information, as well as a couple of API calls for modules to retrieve the version. So now, only version.c will get rebuilt, and the main asterisk binary relinked, which is must faster than rebuilding http.c, manager.c, asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ... The only minor change in behavior here is that the version information reported by chan_sip, for example, is the version of the Asterisk core, and not necessarily the Asterisk version that the chan_sip module came from. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96717 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Print out the name of a function being registered in color, just like the namerussell1-1/+2
of applications when they get registered. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96716 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Merged revisions 96644 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | 2 lines Don't pass an empty string as the device name. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96645 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-04Merged revisions 96575 via svnmerge from tilghman1-9/+17
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines Fix the problem of notification of a device state change to a device with a '-' in the name. Could probably do with a better fix in trunk, but this bug has been open way too long without a better solution. Reported by: stevedavies Patch by: tilghman (Closes issue #9668) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96576 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03Allow the uniqueid to be used for searching for a channel in the list.tilghman1-13/+13
Reported and initially patched by: michael-fig (Closes issue #11340) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96301 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03Compatibility fix for OpenBSDtilghman1-2/+2
Report and fix by: mvanbaak (Closes issue #11669) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96147 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Add doxygen documentation to libresample.h while it's still fresh on my mindrussell1-0/+76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96018 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02For some odd reason, the last set of libresample build changes from Kevin didrussell2-14/+5
not work for everyone, but it did for some. This set of changes makes trunk start again for those having problems. Instead of building libresample as a static library, it just links the object files in directly with the asterisk binary. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95864 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02go back to including libresample in the main Asterisk binary, but this time ↵kpfleming23-1/+10201
including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95816 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02some cleanup of this code while I am trying to debug a problem withrizzo1-15/+19
gdb dying while debugging asterisk. The problem seems to be related with a race in the handling of module_list, which in turn is triggeded by calling dlopen() on a system which uses initializers to create locks. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95772 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Make the translation table show slin16russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95735 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Instead of linking libresample into the main Asterisk binary, build it asrussell22-10195/+1
res_resample, and mark codec_resample as dependent upon res_resample. This prevents the linker from optimizing away libresample, and also makes it so the libresample code isn't linked in to multiple places. (I have another module in a branch that needs it, too.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95697 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merged revisions 95577 via svnmerge from mmichelson1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension into ast_merge_contexts_and_delete (sans the extra lock). (this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the problematic area experienced by the reporters of that issue) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95578 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge changes from team/russell/codec_resamplerussell22-1/+10195
This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Fix -s socket option, and document it as well.qwell1-1/+1
Closes issue #11645, patch by Laureano. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95070 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Merged revisions 95024 via svnmerge from russell1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines Don't report a syntax error when an empty string is passed to ast_get_group. Just return 0. (closes issue #11540) Reported by: tzafrir Patches: group_empty.diff uploaded by tzafrir (license 46) -- slightly changed by me ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95025 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Merged revisions 94977 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec 2007) | 3 lines Fixing a typo in a comment. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94978 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Merged revisions 94828-94829 via svnmerge from russell1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines Change ast_translator_best_choice() to only pay attention to audio formats. This fixes a problem where Asterisk claims that a translation path can not be found for channels involving video. (closes issue #11638) Reported by: cwhuang Tested by: cwhuang Patch suggested by cwhuang, with some additional changes by me. ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines Use the constant that I really meant to use here ... ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94830 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Merged revisions 94824 via svnmerge from kpfleming1-5/+11
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) | 2 lines make this comment explain the situation in an even more explicit fashion ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94825 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Merged revisions 94808 via svnmerge from tilghman1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) | 6 lines Workaround for what is probably a glibc bug (but we'll see this crop up again and again, if we don't add the workaround). Reported by: rolek Patch by: tilghman (Closes issue #11601, closes issue #11426) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94809 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Merged revisions 94801 via svnmerge from russell1-25/+38
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting autoservice, remember it and ensure that the channel has the same setting when autoservice gets stopped. (pointed out by d1mas, patched up by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94802 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Merged revisions 94797 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines When a channel is in autoservice, mark a flag on the channel that says that we only care about the END of a digit. That way, no magic digit emulation stuff will happen when all we're doing is queueing up END frames. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94798 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Leave a note for a minor bug that was pointed out by d1masrussell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94796 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Merged revisions 94790 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines Don't store DTMF BEGIN frames while a channel is in autoservice. It's just going to make ast_read() do a lot of extra work when the channel comes back out of autoservice. (closes issue #11628, patched by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94791 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Adding support for storing the queue log entries in a realtime backend.mmichelson1-9/+25
(closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94782 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-24Merged revisions 94767 via svnmerge from tilghman1-21/+24
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines Race: we need to wait to queue a NewChannel event until after the channel is inserted into the channel list. The reason is because some manager users immediately queue requests from the channel when they see that event and are confused when Asterisk reports no such channel. (Closes issue #11632) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21AST_LIST_REMOVE_CURRENT only takes one argument in trunkmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Merged revisions 94468 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines Since we are freeing list elements within a list traversal, we need to use the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94477 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Merged revisions 94420 via svnmerge from mmichelson1-13/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec 2007) | 5 lines Fixing Portuguese syntax for saying dates and times. Also some coding guidelines cleanup. (closes issue #11599, reported and patched by caio1982, coding guidelines cleanup by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94463 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Merged revisions 94418 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007) | 2 lines Fix for restart-as-user problem reported via the -dev list ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94419 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-20modify http://svn.digium.com/view/asterisk?view=rev&rev=93603rizzo1-60/+84
so that paths and filename are writable by asterisk.c without causing segfaults. This involves defining the variables as const char *, and having them point to as static, writable buffer defined in asterisk.c On passing, fix some errors in using these variables in some files in utils/ , and in res/snmp/agent.c which was redefining a variable without using paths.h (not applicable to 1.4) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94168 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Add a new API function, written at least twice in app_voicemail.crizzo1-0/+37
and likely in other places too. This is quite useful when placing mail/html stuff in config files. /*! \brief Convert some C escape sequences (\b\f\n\r\t) into the equivalent characters. \brief s The string to be converted (will be modified). \return The converted string. */ char *ast_unescape_c(char *s); git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93950 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19add support for PARSE_DOUBLE, and remove identifiers forrizzo1-0/+28
types not supported (INT16 and UINT16) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93925 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Add option for starting remote Asterisk by naming the actual runtime socket ↵oej1-1/+9
instead of pointing to configuration file with -C Reported by: sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax (license 359) doc changes by committer (closes issue #11598) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93854 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Making the canary error message a little more obvious.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93805 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Add a canary process, for high priority mode (asterisk -p) to ensure that iftilghman1-2/+71
Asterisk goes into a busy loop, the machine will be recoverable. We'd still need to do a restart to put Asterisk back into high priority mode, but at least a reboot won't be required. (Closes issue #11559) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93804 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Merged revisions 93625 via svnmerge from mmichelson1-15/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored had one audio file recorded and one empty audio file saved. (closes issue #11529, reported by atis patched by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93626 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18make configuration variable const so they are not accidentallyrizzo1-60/+62
modified. This requires casting the strings in asterisk.c when writing to them, so we do it through a macro to do it consistently. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93603 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18remove unnecessary (char *) casts for ast_config_AST_* variables.rizzo4-4/+4
There are some left in the .flex files, left to the maintainer... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93582 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Rename the macros in defaults.h - they are not meant to berizzo1-14/+15
globally visible. Document the fact that DEFAULT_TMP_DIR cannot be overridden from the default configuration (this needs to be fixed, as you could have a totally different spooldir configured at runtime, and yet DEFAULT_TMP_DIR keeps the compile-time default). Remove two unused entries for sounds and images. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93581 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17Merged revisions 93381 via svnmerge from qwell1-4/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 lines What was I thinking when I wrote this masterpiece? -1 + 1 = 0.. who woulda thunk it?. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93390 f38db490-d61c-443f-a65b-d21fe96a405b
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