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2011-02-15Merged revisions 307879 via svnmerge from rmudgett1-5/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307883 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Fixes bug in chan_sip where nativeformats are not set correctly.dvossel1-0/+8
The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307433 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Pass a MCID request to the bridged channel.rmudgett1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@306755 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.rmudgett1-7/+8
The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306396 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()pabelanger2-6/+5
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306258 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel22-335/+1011
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: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Merged revisions 305040 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31 Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a problem with building the utils directory. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Merged revisions 304950 via svnmerge from tilghman1-25/+36
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) | 18 lines Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used. This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304951 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304245 via svnmerge from mnicholson1-0/+14
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304246 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 303907 via svnmerge from mnicholson1-17/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2 lines Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304152 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-24Merged revisions 303549 via svnmerge from russell1-0/+23
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@303551 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-24According to section 19.1.2 of RFC 3261:mnicholson1-21/+49
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: http://svn.digium.com/svn/asterisk/trunk@303509 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302680 via svnmerge from tilghman1-6/+21
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@302686 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20New astobj2 flag for issuing a callback without locking the container.dvossel1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299135 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Some scheduler API cleanup and improvements.russell4-153/+78
Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299091 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-18Merged revisions 298960 via svnmerge from tilghman3-0/+25
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. Review: https://reviewboard.asterisk.org/r/1055/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298961 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-12Add support for several platforms to obtain the real thread ID.jpeeler2-0/+10
Already had the pthread ID which is not the same. The most obvious enhancement is in the "core show threads" output. As stated in the utils header, if the platform isn't supported -1 is reported (instead of the process ID previously). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298137 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-10Merged revisions 298051 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600 (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) | 11 lines Portability issue on OpenSolaris. Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298052 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-03Merged revisions 297157,297486,297495 via svnmerge from mnicholson1-5/+20
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines Changed some NOTICE and WARNING messages to DEBUG messages. ........ r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines Add support for reserving a fax session before answering the channel. Note: this change breaks ABI compatibility. FAX-217 ........ r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions. Answer the channel before quering it for t.38 support. This is necessary for the query to work properly over local channels. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@297496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 296992 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296992 | tilghman | 2010-12-01 11:01:56 -0600 (Wed, 01 Dec 2010) | 19 lines Merged revisions 296991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600 (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines Clarify documentation on how we store codec preference lists. (closes issue #18397) Reported by: birgita ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296993 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-30Add a comment on why the reserved bit is reserved.tilghman1-1/+7
Came up when reviewing discussion on the CODEC PREFS IE in IAX2. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296826 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revisions 296534 via svnmerge from tilghman1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600 (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines I love standards. There are so many to choose from. Except when there isn't one. Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296535 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295866 via svnmerge from rmudgett2-2/+31
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-20Merged revisions 295711 via svnmerge from russell1-0/+20
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295712 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-18Merged revisions 295441 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295441 | pabelanger | 2010-11-18 13:02:12 -0500 (Thu, 18 Nov 2010) | 11 lines Merged revisions 295440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov 2010) | 4 lines Fix compiler warnings when using openssl-dev 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295442 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Merged revisions 294430 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294430 | tilghman | 2010-11-09 14:33:05 -0600 (Tue, 09 Nov 2010) | 15 lines Merged revisions 294429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@294431 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Merged revisions 294349 via svnmerge from rmudgett1-0/+46
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@294351 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Merged revisions 294278 via svnmerge from jpeeler1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@294279 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Merged revisions 293803 via svnmerge from twilson1-8/+49
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@293809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26Merged revisions 293046 via svnmerge from rmudgett1-30/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) | 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@293047 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Merged revisions 292906 via svnmerge from tilghman1-22/+31
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) | 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@292915 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Merged revisions 291758 via svnmerge from pabelanger1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines Add the ability for ast_find_ourip to return IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@291760 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291192 via svnmerge from dvossel1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines Gtalk enhancements and general code cleanup. This patch includes several chan_gtalk enhancements. Two new gtalk.conf options have been added, externip and stunadd. Setting externip allows us to manually specify what the external IP address is outside of a NAT environment. Setting the stunaddr option to a valid stun server allows for that external ip to be retrieved via a STUN server automatically. This external IP is then advertised during call setup as a possible candidate. I have also attempted to clean up chan_gtalk's code so it meets our coding guidelines. During this cleanup I noticed several things that need to be done in the code and made a TODO section at the top of the file. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@291193 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Merged revisions 290752 via svnmerge from qwell1-31/+22
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290752 | qwell | 2010-10-07 15:58:47 -0500 (Thu, 07 Oct 2010) | 23 lines Merged revisions 290751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@290753 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Merged revisions 290479 via svnmerge from dvossel2-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines Fixes chan_gtalk to work with gmail client This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@290480 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289840 via svnmerge from jpeeler1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289841 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revisions 289543,289581 via svnmerge from tilghman1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@289588 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-28Merged revisions 289104 via svnmerge from tilghman2-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) | 4 lines Solaris compatibility fixes Review: https://reviewboard.asterisk.org/r/942/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288638 via svnmerge from tilghman3-13/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288638 | tilghman | 2010-09-23 22:39:29 -0500 (Thu, 23 Sep 2010) | 16 lines Merged revisions 288637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500 (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines Solaris compatibility fixes ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Merged revisions 287647 via svnmerge from dvossel2-0/+313
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: http://svn.digium.com/svn/asterisk/trunk@287648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Merged revisions 286931 via svnmerge from jpeeler1-4/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286939 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286189 via svnmerge from twilson3-3/+32
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: http://svn.digium.com/svn/asterisk/trunk@286190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 285962 via svnmerge from tilghman1-6/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285962 | tilghman | 2010-09-10 00:32:18 -0500 (Fri, 10 Sep 2010) | 13 lines Merged revisions 285961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines Another fix for Mac OS X. While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285963 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 285931 via svnmerge from tilghman2-27/+43
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines Merged revisions 285930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285932 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-08Merged revisions 285530 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285530 | qwell | 2010-09-08 15:43:10 -0500 (Wed, 08 Sep 2010) | 9 lines Merged revisions 285529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285531 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-07Merged revisions 285336 via svnmerge from tilghman1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010) | 2 lines Fix build on FreeBSD 8.0, take 2. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285337 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-06Merged revisions 285161-285162 via svnmerge from russell1-31/+22
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010) | 4 lines Fix libsrtp -fPIC check for when non-standard prefix is used. Thanks to loompek in #asterisk for reporting the issue and testing this patch. ........ r285162 | russell | 2010-09-06 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line regenerate configure script. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285163 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-04Merged revisions 285057 via svnmerge from russell1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04 Sep 2010) | 2 lines Add a C++ compatible version of AST_CLI_DEFINE(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285058 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03Support for calendar events priorities and categories (with ISO C90 fix)pitel1-0/+2
See RFC 5545 ch. 3.8.1.2 and 9. (closes issue #17837) Review: https://reviewboard.asterisk.org/r/880/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284851 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284610 via svnmerge from tilghman1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
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