asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	fd7d3271c6	Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	cf7131dd6a	Backport state interface changes to app_queue from trunk. After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	df192b77df	Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	23de13f3da	Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0f37862a87	Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	3433f4ecef	use new, improved 8kHz prompts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d64575ac5c	pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	a81abfbd02	Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4a209dbe40	Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	38934ec0d0	Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3e474b3b75	Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	19a67b624c	Fix a crash in IAX2 registration handling found during load testing with dvossel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	2da89b3022	Add missing datastore inherit (exists in all other branches) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f42e9eb6bf	Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e81cde3d9c	Reordering, to change prior to unlocking git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5fc7a5919a	Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	40c5b00494	Export some more required symbols. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	47af9f8fd5	Remove the use of RTLD_NOLOAD, as it is not behaving like expected. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	3c9a8b9aa6	Allow the AES API to work. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	35edd3d0a2	Add missing semicolon in exports script. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dd17912d68	Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	661c6384ad	Allow the CallerID API to work again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	338e48e055	Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	3849208a99	fix typo which broke configure git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c59e2a92d0	Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue 0011261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e536392919	fix another symbol namespace issue (reported by Andrew on asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6efa254bea	Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7e1ee720ba	Improve the build system to properly remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix. With these changes, for a module to export symbols into the global namespace, it must have both the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	59f867a5cb	revert commit that included extranous changes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b8afcedc10	remove accidentally merged properties git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f1f417a9d8	Improve the build system to properly remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix. With these changes, for a module to export symbols into the global namespace, it must have both the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	62fbf19157	Allow dahdichanname to work as advertised. (closes issue #14056) Reported by: dsedivec Patches: load_from_zapata_conf.patch uploaded by dsedivec (license 638) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d09fddd38e	Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3cbc42e2e4	Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a3769669b0	Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	52752df83c	Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	00c1a47c88	Use the correct branch integrated property when generating the version string. Copied the make_version file from Asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8dbfea83ce	Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	1813d4a281	Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	01b90d9092	Fix logic flaw in previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2e825bd259	Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	293900d3f2	Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	e809699406	Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	aedf566905	Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f97c929946	encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	563c72dc84	Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b15b319bd6	Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	21ca773c28	Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. Because using the ast prefix calls are a better choice, ast_free_ptr is the new wrapper for free to pass to functions. Also, a little bit of clean up was done to avoid the debug macros intentionally being redefined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	280153085e	Remove unused variables. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	849820fd54	Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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