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Author	SHA1	Message	Date
Gregory Nietsky	3aa9147fc2	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/branches/10@341580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b761c1eef2	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	69d2d46275	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/branches/10@341486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f62f95c1b4	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	26f196468f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	a4cc7f518e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	aa0c65619d	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	21edfbb5fd	Merged revisions 341312 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fa58ec2c74	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	fe6dd5b23a	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	bbc5c7826e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	a4ce115a56	Set 'core' support level for test_format_api.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	824c5dbfa2	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	db7a2781da	Add information about limitations of new codec support in channel drivers. ... (issue ASTERISK-18680) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f99e9b5bad	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	7a1f2ca356	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	91a65d8b3f	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/branches/10@340971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ad8129697a	Some additional module documentation changes for 10 for the menuselect change. ... (issue ASTERISK-18268) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	cb6e65cdf3	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	d3831204a9	Fixes some support level info so that it can be read by menuselect. ... (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3d8a367788	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e372d326ca	Only send MWI Notify on register if the registration is successful. ... lastmsgssent was removed from chan_sip and the old behavior of sending a mwi notify on register [except when subscribemwi is set] was restored but this must only happen when registration succeeds. leaking information for unsuccessful registrations is not secure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	cdae7c1714	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/branches/10@340718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e00bacf5aa	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	4e1084a392	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	925bcb92b4	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/branches/10@340578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	0482a757c3	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/branches/10@340577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	dc8a317f87	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec1778c05f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	287c621613	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	dbd7f17342	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	baa5ff86e5	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4348fc2cc9	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	3275275a07	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	1e1100cdb4	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/branches/10@340222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	0d1ff9db85	Add a missing file for the astdb2bdb conversion utility ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7ef4224fe6	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f0d579f1d5	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/branches/10@340165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	63d4530e93	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/branches/10@340109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	bc97128a95	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/branches/10@340031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	030a580bb3	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/branches/10@339992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	a1429c5cf6	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/branches/10@339942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	5b1db303a3	Merged revisions 339884 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	f0037b393c	Merged revisions 339830 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	533a924ae5	Merged revisions 339776 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	9926417cac	Reject v17 skinny devices in Asterisk10 ... Small fix for Asterisk10 to reject skinny devices with skinny firmware version17 and above. Review: https://reviewboard.asterisk.org/r/1497/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b1876538ec	Merged revisions 339719 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	6f87c7d755	Fixed segfault on core stop gracefully. ... There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff (in trunk) and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1c6e0d278d	Merged revisions 339625 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	47f22c447b	Merged revisions 339566 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines Update prep_tarball script to download pre-exported documentation. I've updated the prep_tarball script to now download the pre-exported documentation from the Asterisk wiki. This will give us more control over what is being included in the tarball releases, and will make both the PDF and HTML exported documentation look much better (especially when viewing from a console). (Closes issue ASTERISK-18677) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago