asterisk

Commit Graph

Author	SHA1	Message	Date
Gregory Nietsky	b009ea5216	White space fixes in res_fax git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b961d57c4c	Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	0e887d1a50	Fixed typo from previous commit ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5f5e908b19	Updated documentation for the optional CID parameter with CALLERID ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341665 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e2e6e511af	Merged revisions 341599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341599 \| irroot \| 2011-10-20 20:20:08 +0200 (Thu, 20 Oct 2011) \| 8 lines add documentation for check_state_unknown in configs/queues.conf.sample app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	71b7df16bf	Merged revisions 341580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341580 \| irroot \| 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) \| 15 lines Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2f1130e13f	Clean up ast_check_digits The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	3f98c937a1	Merged revisions 341486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341486 \| mnicholson \| 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) \| 18 lines Fix a performance regression introduced in r325483. The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	1ed8cd087a	Outgoing calls with Google Voice Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341436 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5f8648892f	Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	2816ccc516	Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b0076c5be1	Don't resolve numeric hosts or contact unresolved hosts If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10de040b6e	More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	d19ddf8741	Remove an unused include of md5.h Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 Merged-From: http://svn.asterisk.org/svn/asterisk/branches/10@341148 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9f83c2b513	Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	ca0f2acab7	Set 'core' support level for test_format_api.c ........ Merged revisions 341146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2ffea6ddc3	Multiple revisions 341108,341112 ........ r341108 \| pabelanger \| 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) \| 2 lines Voicemail compiler flags are 'core' support ........ r341112 \| pabelanger \| 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) \| 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341122 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	a79c41ee66	Add information about limitations of new codec support in channel drivers. (issue ASTERISK-18680) ........ Merged revisions 341094 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2cb5178d29	Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341089 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	c292e39cdc	Change the internal name of the menuselect options that are used to control whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341023 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	899df042f5	Fix simple switch to not progress a call when call already progressed. If a simple switch was started on a device and then a specific call made (such as redial or speed dial), on timeout of the simple switch the call would be attempted again. This patch only allows the simple switch to make a call if the substate is still in the collecting digits mode. Also added small debug message to dialAndAactivate sub. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4b9546abdf	Merged revisions 340971 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 \| kmoore \| 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) \| 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 \| kmoore \| 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) \| 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e77f1a6ae1	Some additional module documentation changes for 10 for the menuselect change. (issue ASTERISK-18268) ........ Merged revisions 340931 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	19d3e269f6	Avoid unnecessary WARNING message Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340879 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cabf08b511	Fix DTMF blind transfer continuing to execute dialplan after transfer. Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d0ab521e61	Update 10 merged property. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec93d933c0	Restore branch 10 merge properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	3ee015db7a	Merged revisions 339463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 \| irroot \| 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) \| 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	c48bee8e82	Merged revisions 340718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340718 \| schmidts \| 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) \| 9 lines Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 \| schmidts \| 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) \| 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5c77498afd	Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340716 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9d83162d55	Don't skip the query field on a realtime multi query There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340663 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	ee8844782c	Merged revisions 340577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340577 \| schmidts \| 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) \| 9 lines Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 \| schmidts \| 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) \| 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e7ebf7d5ab	Merged revisions 340578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340578 \| twilson \| 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) \| 16 lines Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 \| twilson \| 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) \| 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3bc3e9bbb7	Initialize the PRI channel alarms properly on startup. The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340523 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	796ed62f47	Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340471 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f2cc666a99	Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340419 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9abab10b66	Add protection for SS7 channel allocation and better glare handling. * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340366 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b63c1cc545	Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	1ec8a9d896	Update SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	067250f74c	Convert registered AMI actions to ao2 objects. * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340281 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	15fd1e375c	Return error when no rows are deleted for AMI DBDelTree (closes issue AST-654) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	cf8db24132	Merged revisions 340222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340222 \| twilson \| 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) \| 8 lines On astdb conversion, also warn about permissions requirements The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6708ee76a0	Merged revisions 340219-340220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340219 \| twilson \| 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) \| 8 lines Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ ........ r340220 \| twilson \| 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) \| 2 lines Add a missing file for the astdb2bdb conversion utility ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4ec8d57454	Merged revisions 340165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340165 \| mjordan \| 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) \| 20 lines Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 \| mjordan \| 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) \| 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	bb07ca66a1	Merged revisions 340109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 \| mnicholson \| 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) \| 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 \| mnicholson \| 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) \| 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	7cb2ac8664	Add skinny version 17 protocol support. Added some data to skinny packet structures to make compatible with v17. Added protocolversion to device, set on registration based on the version provided by device. v17 includes some increased ip space for ip6. This patch increases ip space in the packets but still only uses ip4. Some packet structures duplicated (ip4 and ip6 types). ip4 type used unless version is greater or equal to 17. Tested by snuff and myself on 7961 with recent 8.5 firmware. Also tested compatible with old 7960 and older 30VIPs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	c5546e2bd7	Increase SKINNY_MAX_PACKET and add some logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages in v17 that are greater than the old 1000 bytes. Also add some useful logging regarding packet and session handling. A device (with protocol v17) was sending a packet with length greater than 1000 which resulted in the TCP session being destroyed and registration being retryed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	0ac40dc255	Merged revisions 340031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340031 \| wedhorn \| 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) \| 8 lines Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	b90964eda5	Merged revisions 339992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339992 \| wedhorn \| 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) \| 9 lines Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	7e5ce2ac49	Merged revisions 339942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339942 \| igorg \| 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) \| 12 lines Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 \| igorg \| 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) \| 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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