asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	75bb8797f5	Merged revisions 332504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332504 \| kmoore \| 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) \| 15 lines Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 \| kmoore \| 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) \| 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	265102faf8	Merged revisions 332265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 \| rmudgett \| 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) \| 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 \| rmudgett \| 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) \| 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	269082f035	Merged revisions 332119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332119 \| jrose \| 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) \| 23 lines Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 \| jrose \| 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) \| 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ea5262eeb5	Merged revisions 332042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 \| mnicholson \| 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) \| 2 lines fix a code comment AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1858e274e3	Merged revisions 332027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 \| mnicholson \| 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) \| 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 \| mnicholson \| 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) \| 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	4403a89f50	Formatting changes while working with DTMF... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	8f2e8d4b8a	Merged revisions 332022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 \| mnicholson \| 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) \| 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 \| mnicholson \| 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) \| 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3d42d45f25	Merged revisions 331956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331956 \| rmudgett \| 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) \| 20 lines Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 \| rmudgett \| 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) \| 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	30b2f36c72	Merged revisions 331868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331868 \| dvossel \| 2011-08-15 10:14:13 -0500 (Mon, 15 Aug 2011) \| 12 lines Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 \| dvossel \| 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) \| 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	6b7e997df2	Formatting guideline fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	28e2aa76b2	Merged revisions 331772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331772 \| rmudgett \| 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) \| 15 lines Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 \| rmudgett \| 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) \| 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	452f198609	Merged revisions 331715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331715 \| rmudgett \| 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) \| 29 lines Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 \| rmudgett \| 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) \| 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a6ea606a78	Merged revisions 331518 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331518 \| kmoore \| 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) \| 17 lines Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 \| kmoore \| 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) \| 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	dc9513a69d	SIP display-name needed to be empty for Avaya IP500 In order to address a compatability issue with certain features on certain devices which rely on display name content to change behavior, initreqprep in chan_sip.c has been changed to no longer substitute cid_number into the display name when cid_name isn't present. Instead, it will send no display name in that case. (closes issue ASTERISK-16198) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b99b1116be	Merged revisions 331265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 \| rmudgett \| 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) \| 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 \| rmudgett \| 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) \| 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	db3d113414	Merged revisions 330706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330706 \| kmoore \| 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) \| 17 lines Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 \| kmoore \| 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) \| 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	e128ee2567	Merged revisions 330586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330586 \| dvossel \| 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) \| 15 lines Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 \| dvossel \| 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) \| 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	6f112cce0d	Merged revisions 330579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330579 \| dvossel \| 2011-08-02 11:08:57 -0500 (Tue, 02 Aug 2011) \| 9 lines Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 \| dvossel \| 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) \| 2 lines Optimization to buffer initialization fix. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d50e68c827	Merged revisions 330576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330576 \| dvossel \| 2011-08-02 10:55:36 -0500 (Tue, 02 Aug 2011) \| 12 lines Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 \| dvossel \| 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) \| 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	6a15e95a32	astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER. There is a fairly common pattern making its way through the code base where we put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER. The purpose is so that it can be passed into the object hash function. However, this really seems like a hack and potentially error prone. This patch is a first stab at approach to avoid having to do that. It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these situations. Then, the hash function can know whether it was given an object or some custom data to hash. The patch also changes some uses of ao2_find() for iax2_user and iax2_peer objects to reflect how OBJ_KEY would be used. So long, and thanks for all the fish. Review: https://reviewboard.asterisk.org/r/1184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a3ba55e7c7	Merged revisions 330051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330051 \| rmudgett \| 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) \| 29 lines Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 \| rmudgett \| 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) \| 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 \| rmudgett \| 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) \| 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	16a32f5030	Merged revisions 329995 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329995 \| qwell \| 2011-07-28 10:45:49 -0500 (Thu, 28 Jul 2011) \| 13 lines Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 \| qwell \| 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) \| 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	73462b32dd	Merged revisions 329896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329896 \| seanbright \| 2011-07-28 07:35:27 -0400 (Thu, 28 Jul 2011) \| 9 lines Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 \| seanbright \| 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) \| 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	5c627eba2b	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Russell Bryant	f243d129c9	Merged revisions 329257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 \| russell \| 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) \| 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	54f92a68c7	Merged revisions 329204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329204 \| rmudgett \| 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) \| 13 lines Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 \| rmudgett \| 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) \| 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	9c232a5470	Merged revisions 328936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328936 \| kmoore \| 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) \| 15 lines Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 \| kmoore \| 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) \| 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1dc97eb69b	Merged revisions 328824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 \| kmoore \| 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) \| 18 lines Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 \| kmoore \| 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) \| 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	8888df3a23	Merged revisions 328611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328611 \| markm \| 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) \| 15 lines Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 \| markm \| 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) \| 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	a0573368a1	Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc. (closes issue ASTERISK-17947) Review: https://reviewboard.asterisk.org/r/1239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	145c174565	Merged revisions 328329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 \| rmudgett \| 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) \| 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4a7726b605	Merged revisions 328317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328317 \| rmudgett \| 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) \| 13 lines Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 \| rmudgett \| 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) \| 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 \| lmadsen \| 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) \| 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3b4d9075f6	Merged revisions 327682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327682 \| twilson \| 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) \| 9 lines Update chan_gtalk to work with changed GMail-based calls The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e613fd544	Merged revisions 327211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 \| rmudgett \| 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) \| 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1353e83531	Merged revisions 327044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327044 \| russell \| 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) \| 2 lines Resolve some set-but-unused-variable warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	ba1cc98f1a	Merged revisions 326683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 \| mnicholson \| 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) \| 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	14553512ee	Merged revisions 326681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 \| mnicholson \| 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) \| 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a7c6f0445e	Fixes newlines from being stripped from out of dialog sip MESSAGES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 \| tilghman \| 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) \| 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	14d510c5b7	Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 \| rmudgett \| 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) \| 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	76e4e2e777	Merged revisions 326144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 \| rmudgett \| 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) \| 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	356e18629b	Fixes warning message caused by confbridge playback chan not being answered. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	39a7152df3	Merged revisions 325935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 \| rmudgett \| 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) \| 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	1d93d217f0	Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 \| kmoore \| 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) \| 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	37d6d89d97	Merged revisions 325416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 \| kpfleming \| 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) \| 3 lines Fix random misspelling noticed on asterisk-users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bb4e0c7f7c	Merged revisions 325339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 \| dvossel \| 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) \| 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	70be58c1a7	Merged revisions 325212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 \| rmudgett \| 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) \| 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4812697542	Fixes issue with video and text not being reinvited correctly with directmedia If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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