asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	4731bae48d	Breakage from slightly before the outage; would have fixed sooner but for the outage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	268318bf88	Backport the use of curl from 1.6.2 to make the 1.4 target work on Bamboo. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	17ca06cdb8	Partial revert of r315671 which removed a logging statement and not a manager event. Reported by ibercom in #asterisk-bugs. (issue #16033) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	acfd6d4bad	Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a851a4834f	Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	77acf5b626	Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	dac18ad759	Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	112e8ef678	Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	94fa98557d	Reverted part of r314607, as it can introduce a regression. Specifically, the security check for the "system" privilege was removed. If a user had the "call" privilege but not the "system" privilege, they would loose the ability to execute the system app and dialplan functions that run commands in a shell. This branch never used the "system" privilege for that purpose and did not need to be patched. AST-2011-006 (related to issue 0018787) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	b3d58b6260	chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	1cfde1088c	Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5a3d415ea4	Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9bd88702b7	Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	05ea303ca4	app_voicemail: Fix ODBC Storage compile regression caused by me, from mantis bug #19032 / commit r312070 (closes issue #19142) Reported by: vrban Patches: app_voicemail_fix_for_312070.patch uploaded by vrban (license 756) Tested by: vrban, alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	fa67a98ccc	Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	453275054e	Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	afbfce2971	Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	059e1a2dec	Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	cd24c868cb	Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	e7597d55df	app_voicemail: leave_vociemail doesn't use last_message_index to store next message trivial change to bring inline with 1.6.2 1.8 and trunk. The symptom was if msg0000 was missing, and the last was msg0004, the next msgnum would be msg0000 when it should have been msg0005 (issue #18998) Reported by: tootai Patches: bug18998.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f6cd305277	Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	4de07be466	voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	752c3ef591	app_voicemail:close_mailbox imap_storage doesn't use last_msg_index git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	c875064714	app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	c0e4bdd609	Don't hold the pvt lock while streaming a file. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9dbd90f447	Don't dec the usecount of an eventqent then use it. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	5510659c1f	Remove the provisional keepalive scheduler entry's reference to the pvt when we remove the scheduler entry. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	93d2033f4b	Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	bf061fa778	Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	ba3c804b06	core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	5e8dfa3e6b	"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	75dc421e15	Add AELSub, which provides a stable entry point into AEL subroutines. This commit needs some explanation, given that we're adding a new application into an existing release branch. This is generally a violation of our release policy, except in very limited circumstances, and I believe this is one of those circumstances. The problem that this solves is one of the sanity of using multiple dialplan languages to define a dialplan. In the case of the reporter, he or she is using AEL is define subroutines, while using Realtime extensions to invoke those subroutines. While you can do this, it's based upon the reality of AEL using actual dialplan extensions; however, there is no guarantee that the details of _how_ AEL is compiled into extensions will remain stable. In fact, at the time of this commit, it has already changed twice, once in a fundamental way. Now normally, a new application would only be added to trunk. However, this application is explicitly to create a stable user-level API between versions, and adding it to trunk only will not solve the user's problem of switching between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10. Therefore, it needs to go into existing release branches. For the sake of consistency, and also because one of the changes was between 1.4 and 1.6.x, I am also electing to commit this to 1.4. (closes issue #18910) Reported by: alexandrekeller Patches: 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14) 20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14) Tested by: alexandrekeller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	eafdcbacd4	Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	744b5c4bbb	Don't try to free statically allocated memory. Note: compiling after ./configure --enable-dev-mode will keep these kinds of mistakes from being committed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jonathan Rose	aaf50c4688	Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Ruggles	fd2d1c2139	fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	e90101cecf	Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	ebc809b6d4	silence gcc 4.2 compiler warning git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2e714923c3	Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	788e66cf20	Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	212f41ddf3	Reentrancy problem if outgoing call gets different B channel than requested. The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk channel with the channel lock when it changes the technology private pointer to a new private structure. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. Partial backport from v1.8 -r300714. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@307623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	af36e55080	Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'll be reverting shortly. (issue #18776) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@307534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	9b490b1adc	Fix comparison for REFER Replaces tags with pedantic=yes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	d6eabdc0dd	fix this line again git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	fb475c8c7e	clean this up, sorry my brain is not really working git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8bd2bfb587	Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e9dcd4a9e2	make this safer and fully correct, pointed out by Steve Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	dceced3626	Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	c80738307a	Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a7d12781c0	Fix no MOH and frame queueing problem for parked calls. This was a regression introduced when select was changed to poll and was just a conversion error: POLLPRI detects OOB data, not POLLERR. (closes issue #18637) Reported by: jvandal git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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