asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	f8f032e98b	Blocked revisions 173248 via svnmerge ........ r173248 \| dvossel \| 2009-02-03 17:35:55 -0600 (Tue, 03 Feb 2009) \| 6 lines Fixes issue with IAX2 transfer not handing off calls. Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	e70fc3d136	Broke up the large conditional blocks so it is easy to see if a function is compiled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e179e613f7	1. Make OS X compile cleanly with app_stack. 2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	673d85387a	Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 \| tilghman \| 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) \| 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	34be09bf5c	Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 \| twilson \| 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) \| 2 lines Fix a feature inheritance bug I added after code review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c668cbfbfc	Reverting commit number 173028 as there are some potential issues git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7db67f9ca7	Add a CLI command to log out a manager user (closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	8be58b59c1	Recorded merge of revisions 172962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172962 \| rmudgett \| 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) \| 11 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	53d9b77898	This reverts the changes I made for 11583; will	17 years ago
Leif Madsen	fdcc0a9a60	Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	c61e8a7865	This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	326587ebe3	Fix a spelling mistake. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	a9ee30da54	Add a todo. I do need to really check what's going on with this kill-the-user business ;-) Why do we suddenly have two flags to set peer type? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	81a3d40c08	Small formatting change git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	9ea148b260	Add some well-needed improvements to the wishlist in the code, so that we can close some bug reports. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2af8f59958	The CID lookup feature wasn't actually working properly with dialog-info+xml supporting devices. The devices (snoms, specifically) need to receive a SIP URI instead of just an extension. This adds that functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	317e4e7313	Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a2ddc0bb5e	Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	27862b0dce	Blocked revisions 172639 via svnmerge ........ r172639 \| twilson \| 2009-01-30 18:15:09 -0600 (Fri, 30 Jan 2009) \| 5 lines Rename new parkedcallparking option to parkedcallreparking Since this option actually already existed in 1.6.0+, use the same name so as not to confuse people when they upgrade ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	458dfe4748	Fix redefinition of flag in channel.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	a010aa5ade	Remove incorrect line from sample config git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	8d782f96b8	Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 \| twilson \| 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) \| 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	402c61117a	Parameter position reversed in documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	80f91f6ab7	Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 \| tilghman \| 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) \| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3b08de23df	Remove tabs from comment git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	97b4e9cf2a	channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	14cebb2c38	Fixed some doxygen comments git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	efd6586546	Revert two lines that was extra, but only on fridays. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	7ecda45482	Fix "cancel answered elsewhere" through app_queue with members in chan_local. Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b3ab95317c	Better document mode=multirow, based upon a conversation with Jared. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	898fd64528	The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	0685c4b281	Update documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	b79a12e929	- Make sure we set setvar= variables on outbound calls too, not only inbound calls. - Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	3e144e2a71	Make sure register= line supports both port and expiry at the same time. (closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	55782a8dfa	Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 \| oej \| 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) \| 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f029e410ec	A further correction: cast the sizeof to an int. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	47db0f64f2	Fix how we skip fields (to avoid fields which don't exist) when doing an UPDATE. (closes issue #14205) Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5918a38b98	my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	268ac221a2	Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 \| murf \| 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) \| 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ca052f64d0	Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 \| tilghman \| 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) \| 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	c61e33b927	Yep. Documentation is important. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	097822966b	Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on. This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	aca43d126a	Add some more notes about device matching. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2c4f19eb2c	Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 \| oej \| 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) \| 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dda3fd446f	Fix some signedness problems in func_aes.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	7a69506727	Don't complain about lack of D-channels on PTMP connections git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	abf70664ab	Adding AES_ENCRYPT and AES_DECRYPT dialplan functions. (closes issue #14301) Reported by: amorsen review: http://reviewboard.digium.com/r/128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5f1a4ebe6d	Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 \| mmichelson \| 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) \| 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	fc7455fa44	Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 \| mmichelson \| 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) \| 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	04e56bde03	Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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