asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	786f5898d1	Finalize ast_channel opaquification Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	a9d607a357	Opaquify ast_channel structs and lists Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	23bc964e1c	Constify some more channel driver technology callback parameters. Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	590391c038	Fix typo from r333070 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	82e1dda364	Formatting changes - Removing some red white space and adding some curly brackets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	64fb851843	Add manager event for local channel semi-bridge (issue AST-17623) Review: https://reviewboard.asterisk.org/r/1154 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	4403a89f50	Formatting changes while working with DTMF... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 \| lmadsen \| 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) \| 14 lines 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/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	34e2305ae7	Merged revisions 324048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 \| twilson \| 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) \| 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	3588746c75	Merged revisions 321515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 \| dvossel \| 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) \| 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	bb5e875b65	Merged revisions 316330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316330 \| dvossel \| 2011-05-03 16:37:59 -0500 (Tue, 03 May 2011) \| 24 lines Merged revisions 316329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316329 \| dvossel \| 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) \| 17 lines Merged revisions 316328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 \| dvossel \| 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) \| 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	83ad7a9e6c	Merged revisions 315446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315446 \| russell \| 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) \| 14 lines chan_local: resolve a deadlock. This patch resolves a fairly complex deadlock that can occur with the combination of chan_local and a dialplan switch, such as dynamic realtime extensions, which pulls autoservice into the picture when doing a dialplan lookup. (closes issue #18818) Reported by: nic Patches: issue18818.patch uploaded by jthurman (license 614) 18818.v1.txt uploaded by russell (license 2) Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	472e9aca3f	Merged revisions 315053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315053 \| alecdavis \| 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) \| 23 lines Merged revisions 315052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315052 \| alecdavis \| 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) \| 16 lines Merged revisions 315051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315051 \| alecdavis \| 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) \| 11 lines chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	36da6b6286	Merged revisions 306127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 \| twilson \| 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) \| 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 \| twilson \| 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) \| 16 lines Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 \| twilson \| 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) \| 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 years ago
Sean Bright	d42cb6fd1d	Merged revisions 302412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 \| seanbright \| 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) \| 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ac87fc136d	Merged revisions 299626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299626 \| tilghman \| 2010-12-25 04:07:15 -0600 (Sat, 25 Dec 2010) \| 19 lines Merged revisions 299625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299625 \| tilghman \| 2010-12-25 04:05:00 -0600 (Sat, 25 Dec 2010) \| 12 lines Merged revisions 299624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 \| tilghman \| 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) \| 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	7189a944be	Merged revisions 292868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292868 \| dvossel \| 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) \| 39 lines Merged revisions 292867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 \| dvossel \| 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) \| 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 \| dvossel \| 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) \| 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4e473de5e2	Merged revisions 288748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288748 \| twilson \| 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) \| 19 lines Merged revisions 288747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288747 \| twilson \| 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) \| 12 lines Merged revisions 288746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288746 \| twilson \| 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) \| 5 lines Don't fail a masquerade if it is already being hung up This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5ad9625cbf	Merged revisions 288507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288507 \| twilson \| 2010-09-22 16:18:27 -0700 (Wed, 22 Sep 2010) \| 22 lines Merged revisions 288500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288500 \| twilson \| 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) \| 15 lines Merged revisions 288499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 \| twilson \| 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) \| 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288519 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
Jeff Peeler	a0460f3b9c	Merged revisions 281466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281466 \| jpeeler \| 2010-08-09 18:04:02 -0500 (Mon, 09 Aug 2010) \| 2 lines Add some more stuff to copy from 281429. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	416b05e9da	Merged revisions 281429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281429 \| jpeeler \| 2010-08-09 15:43:54 -0500 (Mon, 09 Aug 2010) \| 27 lines Merged revisions 281391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281391 \| jpeeler \| 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) \| 20 lines Merged revisions 281390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281390 \| jpeeler \| 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) \| 13 lines Prevent loss of Caller ID information set on local channel after masquerade. Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	3def1196b4	Merged revisions 280307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280307 \| mnicholson \| 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) \| 11 lines Merged revisions 280306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 \| mnicholson \| 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) \| 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280308 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
Russell Bryant	d27bb2d811	Only call ast_channel_cc_params_init() if allocating a channel succeeds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278051 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	f4d96da591	Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 \| tilghman \| 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) \| 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tim Ringenbach	c6b7eae5e6	Add new AMI command LocalOptimizeAway. This command lets you request a "/n" local channel optimize itself out of the way anyway. Review: https://reviewboard.asterisk.org/r/732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	6722251986	Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 \| dvossel \| 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) \| 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	7509949658	Prevent crash when originating a call to a local channel. Call completion code tries to grab the call completion parameters from the requesting channel during local_request. When originating a call to a local channel, however, this channel is NULL. This was causing an issue for me when trying to run a test script. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	df3fc304c9	Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 \| russell \| 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) \| 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	73ef4b8daf	Removed cdrflags from ast_channel structure. Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	862ebf4d00	fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	acd243ca65	Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 \| jpeeler \| 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) \| 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	556260ad93	Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/9700@default&Local/0009700@default) exten => 9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => 9700,n,wait(3) ;3 works, 1 did not exten => 9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	72c1b76038	Merged revisions 244070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244070 \| tilghman \| 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) \| 16 lines Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue. (closes issue #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt uploaded by tilghman (license 14) 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: kobaz, atis (closes issue #16581) Reported by: ZX81 (closes issue #16681) Reported by: alexr1 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	91a45e4d3e	Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 \| tilghman \| 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) \| 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7a01655732	Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 \| tilghman \| 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) \| 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	85dd68ca7a	Merged revisions 230038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230038 \| file \| 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) \| 9 lines Fix a crash caused by two threads thinking they should both free the chan_local private structure when only one should. (closes issue #15314) Reported by: sroberts Patches: Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780) Tested by: davidw, lottc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	d6b5df8715	Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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