asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	ed72dd52a8	Merged revisions 331575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331575 \| rmudgett \| 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) \| 9 lines Segfault in shell_helper in func_shell.c. The return value of popen() was not checked for failure to open. (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	89b58dc234	Merged revisions 328540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328540 \| tilghman \| 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) \| 2 lines Typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 \| tilghman \| 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) \| 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e789eb8b2d	CHANNEL(pickupgroup) Allow Setting / Reading the pickupgroup of a channel with func_channel.c (closes issue #19045) Reported by: irroot Review: https://reviewboard.asterisk.org/r/1148/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1b57da8673	Adds STRREPLACE function Adds a new STRREPLACe function to func_strings.c that allows users to search and replace against a variable in the dialplan. (closes issue #18023) Reported by: wdoekes Review: https://reviewboard.asterisk.org/r/1219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	00dc1556ab	Fixes reliability issues with func_jitterbuffer's usage in the new ConfBridge application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	1fca95b1d4	Merged revisions 316094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316094 \| tilghman \| 2011-05-02 14:09:55 -0500 (Mon, 02 May 2011) \| 15 lines Merged revisions 316093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316093 \| tilghman \| 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) \| 8 lines More possible crashes based upon invalid inputs. (closes issue #18161) Reported by: wdoekes Patches: 20110301__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	18d591cb48	Introduction of the JITTERBUFFER dialplan function. Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	fd2f95b3c4	Merged revisions 314206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314206 \| lmadsen \| 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) \| 14 lines Merged revisions 314205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314205 \| lmadsen \| 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) \| 6 lines Remove duplicate documentation from func_channel.c (closes issue #18970) Reported by: IgorG Patches: func_channel.c.doc.diff uploaded by IgorG (license 20) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1c1c9c2bd4	Merged revisions 310587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310587 \| jrose \| 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) \| 15 lines Merged revisions 310585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310585 \| jrose \| 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) \| 8 lines Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced. When it is off, DTMF will not be processed by the function. Programmed by Jonathan Rose Reviewed by David Vossel, Leif Madsen, and Russell Bryant http://reviewboard.digium.internal/r/93/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	9650fb3e1a	Merged revisions 310415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310415 \| tilghman \| 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) \| 14 lines Merged revisions 310414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310414 \| tilghman \| 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) \| 7 lines Transactional handles should be used for the insertbuf, if available. Also, fix a possible resource leak. (closes issue #18943) Reported by: irroot ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	67c91388db	Merged revisions 310142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310142 \| tilghman \| 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) \| 19 lines Merged revisions 310141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 \| tilghman \| 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) \| 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 \| tilghman \| 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) \| 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 \| rmudgett \| 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) \| 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ffe9e4acfc	Merged revisions 309170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309170 \| rmudgett \| 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) \| 7 lines Document CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Added XML documentation for CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML documentation for CHANNEL(reversecharge). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	008aa0e3b8	Merged revisions 308991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308991 \| tilghman \| 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) \| 14 lines Merged revisions 308990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308990 \| tilghman \| 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) \| 7 lines Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled. (closes issue #18815) Reported by: irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot (license 52) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	e38fa2d3cd	Merged revisions 307837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307837 \| tilghman \| 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) \| 15 lines Merged revisions 307836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 \| tilghman \| 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) \| 8 lines Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	2740326200	Merged revisions 305844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305844 \| tilghman \| 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) \| 5 lines Eliminate a file descriptor leak when using the FILE() dialplan function. (closes issue #18731) Reported by: marioabajo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	93bade5639	Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	f9c3b26241	Add Function and Application Relationships to documentation Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e706b5706e	According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	52dbebad8e	Add DB_KEYS. Discussion on #asterisk on 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) leifmadsen: you need to know the tree structure already though (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan (02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything (02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script (02:10:13 PM) boch: for example i need to know all entries in the tree (02:10:15 PM) boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) leifmadsen: that's the problem (02:10:22 PM) boch: thank you (02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. (02:15:35 PM) leifmadsen: database shows everything (02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>' (02:16:41 PM) leifmadsen: guess no one has found that important enough to program :) (02:16:52 PM) leifmadsen: at that point you should probably just use a relational database... (02:17:10 PM) mateu: i dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does (02:17:24 PM) leifmadsen: no one has programmed it though (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen (02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS() (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family? (02:34:02 PM) leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate through layers of them? (02:34:30 PM) leifmadsen: i.e. family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, yes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	7cb1c06dd3	Add relationships to function documentation. Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	ca8a5498b1	Add relationships to function documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b2a70b4065	Oops, missed the actual decoding part. (closes issue #18046) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a58b2fb395	XML validation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	473e176df8	Add a hashcompat mode called "legacy", which translates a literal plus sign to a space. (closes issue #18046) Reported by: wdoekes Patches: 20100930__issue18046.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ac77932bac	Merged revisions 298478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298478 \| tilghman \| 2010-12-16 02:56:13 -0600 (Thu, 16 Dec 2010) \| 15 lines Merged revisions 298477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298477 \| tilghman \| 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) \| 8 lines Eliminate duplicates from container. (closes issue #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt uploaded by tilghman (license 14) Tested by: bunny ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	53357354a4	Merged revisions 294989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294989 \| tilghman \| 2010-11-15 01:44:38 -0600 (Mon, 15 Nov 2010) \| 15 lines Merged revisions 294988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 \| tilghman \| 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) \| 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	34c30c8ad3	Merged revisions 293159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293159 \| jpeeler \| 2010-10-28 11:11:08 -0500 (Thu, 28 Oct 2010) \| 18 lines Merged revisions 293158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 \| jpeeler \| 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) \| 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6d0e383321	Merged revisions 289543,289581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289543 \| tilghman \| 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) \| 2 lines More Solaris compatibility fixes ........ r289581 \| tilghman \| 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) \| 2 lines Solaris fixes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	794ff358a3	Merged revisions 288713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288713 \| tilghman \| 2010-09-24 08:54:17 -0500 (Fri, 24 Sep 2010) \| 12 lines Merged revisions 288712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288712 \| tilghman \| 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) \| 5 lines Solaris won't printf a NULL. (closes issue #18041) Reported by: asgaroth ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2f3dee2379	Merged revisions 287647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287647 \| dvossel \| 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) \| 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d04046fbe7	Merged revisions 286189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286189 \| twilson \| 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) \| 30 lines Merged revisions 286115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 \| twilson \| 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) \| 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 \| twilson \| 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) \| 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1c12ca0407	Merged revisions 285484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285484 \| tilghman \| 2010-09-08 02:14:17 -0500 (Wed, 08 Sep 2010) \| 2 lines Documentation only ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2302618bb7	Merged revisions 285373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285373 \| tilghman \| 2010-09-07 16:14:03 -0500 (Tue, 07 Sep 2010) \| 7 lines Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up. (closes issue #17652) Reported by: kobaz Patches: func_channel.patch uploaded by kobaz (license 834) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8190e96fad	Merged revisions 284610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 \| tilghman \| 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) \| 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2de5bbc89f	Merged revisions 283350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283350 \| russell \| 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) \| 2 lines Don't attempt to release a NULL ODBC handle. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	42490d744b	Merged revisions 280809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280809 \| tilghman \| 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) \| 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d6e1c724e5	Remove built-in AES code and use optional_api instead Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	0ae9097e3e	Oops, XML documentation fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fc9efc4ff5	It really cannot fail in the places below, but the stupid compiler doesn't know that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e939dfea9d	Weird compiler error on Bamboo. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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578 Commits (c69f237d3d83c24d1d41d9eb16c5fee14099d053)