summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Merged revisions 195991 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Pass connected line updates along during a bridge.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Rework the cdr_custom.conf.sample header a bit to reflect the changes inSean Bright
functionality (allowing multiple mappings). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Merged revisions 195881 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20If a variable had a blank value upon the initial setting, then it would do ↵Tilghman Lesher
nothing. Identified by Dmitry Andrianov via private email, fixed by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20Get rid of some duplicated code and correct a connected line error.Mark Michelson
When receiving a 200 OK response to an INVITE, it was possible to transmit two connected line updates instead of a single one. Furthermore, the second did not have the proper information present. Now the two have been combined into a single update and the correct information is presented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20Plug a memory leak in app_dial.Mark Michelson
Since we may have copied connected line info into the chanlist struct prior to placing an outbound call, we need to be sure to free the allocated data when we hang the call up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20Merged revisions 195688 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20Merged revisions 195635 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19Add basic support for handling connected line-related UPDATE requests.Mark Michelson
SIP purists may want to look the other way... When COLP/CONP support for SIP was committed, there was a condition under which Asterisk may transmit a SIP UPDATE in order to communicate the change in connected line information. The issue here is that while we could send a SIP UPDATE message, we were not prepared to receive such an UPDATE and would always responde with a 501 when we received an UPDATE. The situation was a bit rough. We really want to be able to receive UPDATEs having to do with connected line changes, but the amount of effort involved in properly supporting RFC 3311 was staggering. This commit represents a compromise. First, it was decided that it is important to only send a SIP UPDATE to an endpoint that is able to handle one. So, now we have added parsing of the Allow header into SIP. We store the allowed methods on SIP peers so that when we communicate with them, we already will know what we can and cannot send to them. We will parse the peer's allowed methods when he registers with us. If the peer is not the type to register with us, but the qualify option is enabled, then we will use the response to the OPTIONS request we send the peer to determine the peer's allowed methods. When the peer's registration expires, or when qualify deems the peer to be unreachable, we clear the allowed methods from the peer. For an actual call, we will copy the peer's allowed methods to the sip_pvt representing the call leg. If we are communicating with an endpoint which is not a peer, then we will just parse the Allow header from the first message we receive during the call and store the information in the sip_pvt. If, during communication with a peer, we receive a 501 response, then we will make sure to save the fact that we cannot use that method when communicating with that peer. Now, with all that infrastructure in place, the only actual place we use this information currently is when attempting to send a connected line change using an UPDATE request. If we cannot send the change immediately using an UPDATE, we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon as it is allowed. The second part of the changes here is for Asterisk to accept UPDATE requests that have connected line changes. Since we are not fully supporting RFC 3311, Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead, if you are communicating with what you know to be another Asterisk box, you may set the rpid_update parameter in sip.conf so that we will send UPDATEs to that Asterisk box. When we send a connected line update, we set a custom header called "X-Asterisk-rpid-update." On the receiving end, if Asterisk receives an UPDATE that does not have the "X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501 since media-changing UPDATEs are not supported. We should never get such UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow header. If the custom header is present in the received UPDATE, though, then we will check the incoming request for connected line updates and queue the update on the channel where the change occurred. ABE-1840 ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19Merged revisions 195520 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19Merged revisions 195448 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Recorded merge of revisions 195366 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Fix the CLI command 'manager show command' documentation and functionality.Eliel C. Sardanons
The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Rollback commit 195367.Eliel C. Sardanons
The CLI command 'manager show command' supports passing multiple AMI actions at a time. The issue with this command was in another place. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Avoid autocompleting passed the action name argument in the CLI command.Eliel C. Sardanons
When running the autocomplete of the CLI command 'manager show command <action>' it was autocompleting everything else after the <action> argument, giving an error, because this command doesn't support multiple AMI action names at a time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Move AGI documentation from static to the XML form.Eliel C. Sardanons
Move the AGI commands 'receive text', 'receive char' and 'record' static documentation to XML docs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Move the spawn of astcanary down, until after the call to daemon(3).Tilghman Lesher
This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Fix externalivr's setvariable command so that it properly sets multiple ↵Mark Michelson
variables. The command had a for loop that was guaranteed to only execute once since the continuation operation of the loop would set the input buffer NULL. I rewrote the loop so that its operation was more obvious, and it would set multiple variables correctly. I also reduced stack space required for the function, constified the input string, and modified the function so that it would not modify the input string while I was at it. (closes issue #15114) Reported by: chris-mac Patches: 15114.patch uploaded by mmichelson (license 60) Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Remove some unused code.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18The facilityenable parameter does not have anything to do with pritimer ↵Richard Mudgett
parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Const-ify a string, fix a log message, and use the correct signature for theSean Bright
load_module function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Merged revisions 195206 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present. (closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Allow cdr_custom to write to multiple files instead of just one.Sean Bright
Up to now, cdr_custom would only accept a single filename/format from cdr_custom.conf. This change allows you to specify multiple filename & format directives. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Warn about the use of the application WaitExten() within a Macro().Eliel C. Sardanons
Update applications documentation to warn the user about the use of the WaitExten() application within a Macro(). Recommend the use of Read() instead. (closes issue #14444) Reported by: ewieling git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Merged revisions 195095 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited. (closes issue #13569) Reported by: bkw918 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Fix a bug where specifying an empty outboundproxy would cause packets to get ↵Joshua Colp
sent to ourself. (closes issue #15106) Reported by: timeshell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Do not avoid loading the XML documentation if not XInclude substitution is done.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Recorded merge of revisions 195020 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) | 5 lines Don't try to unlock a bogus channel. (closes issue #15144) Reported by: cristiandimache ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-16Allow to include sections of other parts of the xml documentation.Eliel C. Sardanons
Avoid duplicating xml documentation by allowing to include other parts of the xml documentation using XInclude. Example: <xi:include xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" /> (Insert this line to include the synopsis of the CHANNEL function xml documentation). It is also possible to include documentation from other files in the 'documentation/' directory using the href="" attribute inside a xinclude element. (closes issue #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-16Fix a missing unlock in case of error, and a missing free().Eliel C. Sardanons
Always free the allocated memory for a string field, because we are always using it (not only when xmldocs are enabled). Also if there is an error allocating memory for the string field remember to unlock the list of registered applications, before returning. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Merged revisions 194873 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Merged revisions 194557,194685 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Merged revisions 194764 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Shuttle some bits around to address some gain issues with G.722.Russell Bryant
(closes AST-209) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Further simplify codec_g722 build.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Actually force running make for g722.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15add elielMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Allow to specify an enumlist inside an enum.Eliel C. Sardanons
It was not possible to use an enumlist inside an enum: <enumlist> <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist> Now we will be able to insert as many levels as we want. (closes issue #15112) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Add ability for modules to dynamically register logger levelsKevin P. Fleming
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf. Review: https://reviewboard.asterisk.org/r/244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Merged revisions 194509 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May 2009) | 1 line Update URL to Reviewboard ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Merged revisions 194484 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Add outgoing_colp misdn.conf port parameter.Richard Mudgett
Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Fix a typo where an equality check should be an assignment.Russell Bryant
(closes issue #15103) Reported by: lmsteffan Patches: transfer_crash.patch uploaded by lmsteffan (license 779) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Fix a bug where the 'T' option to Meetme did not work.Joshua Colp
(closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14If the timing ended on a zero, then we would loop forever.Tilghman Lesher
(closes issue #14983) Reported by: teox Patches: 20090513__issue14983.diff.txt uploaded by tilghman (license 14) Tested by: teox git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-13Blocked revisions 194356 via svnmergeMark Michelson
........ r194356 | mmichelson | 2009-05-13 14:41:44 -0500 (Wed, 13 May 2009) | 13 lines Remove an extraneous unlocking operation from ast_channel_free. In the case that we could not remove the desired channel from the list of channels, there was an extra call to unlock the channel list. Since we unlock the list later on in the function anyway, this results in the list being unlocked twice yet only being locked once. (closes issue #15098) Reported by: tim_ringenbach Patches: remove_extra_unlock.diff uploaded by tim (license 540) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-13Do not lock the 'sessions' container, lock the allocated 'session'.Eliel C. Sardanons
There was a typo in the structure being locked, and we were locking the 'sessions' container instead of the 'session' structure thar we are modifying. Reported by seanbright on #asterisk-dev, thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-13Merged revisions 194208 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-06 22:23:21 +0000